Re: [asterisk-users] async agi question
On Mon, Apr 13, 2009 at 6:59 AM, wrote: > Hi Moy, > > thanks a lot for your fix, but I'm afraid it doesn't work. I looked your > patch over and I realize the code never passes by neither of the two lines > you added with "returnstatus = AGI_RESULT_HANGUP". Even, it seems the > execution doesn't pass by res_agi.c at all, or at least, it doesn't pass over > any "ast_log(LOG_DEBUG,..." lines like the ones your last patch has above the > returnstatus fix. Could be the execution is flowing down by an "if - else - > break" without an "ast_log(LOG_DEBUG,..." line? In that case, would the > "returnstatus = AGI_RESULT_HANGUP" be added to any places more? > > Below is the output log for the redirect while playing a file. As you can > see, there isn't any res_agi.c output on it: > > [Apr 13 11:20:09] DEBUG[5804]: manager.c:2108 process_message: Manager > received command 'Redirect' > [Apr 13 11:20:09] DEBUG[5804]: channel.c:1378 ast_softhangup_nolock: > Soft-Hanging up channel 'SIP/501-08287a00' > [Apr 13 11:20:09] DEBUG[5815]: channel.c:1793 ast_settimeout: Scheduling > timer at 0 sample intervals > [Apr 13 11:20:09] DEBUG[5815]: pbx.c:2448 __ast_pbx_run: Extension 801, > priority 0 returned normally even though call was hung up > [Apr 13 11:20:09] DEBUG[5815]: channel.c:1378 ast_softhangup_nolock: > Soft-Hanging up channel 'SIP/501-08287a00' > [Apr 13 11:20:09] DEBUG[5815]: channel.c:1477 ast_hangup: Hanging up channel > 'SIP/501-08287a00' > [Apr 13 11:20:09] DEBUG[5815]: chan_sip.c:3485 sip_hangup: Hangup call > SIP/501-08287a00, SIP callid 2dbe6797392cde921fb7db0b16e81...@10.0.5.20) > > However, if the redirect is done without playing a file, the execution does > pass by res_agi.c: > > [Apr 13 12:03:57] DEBUG[2688]: manager.c:2108 process_message: Manager > received command 'Redirect' > [Apr 13 12:03:57] DEBUG[2688]: channel.c:1378 ast_softhangup_nolock: > Soft-Hanging up channel 'SIP/501-08279028' > [Apr 13 12:03:57] DEBUG[2755]: res_agi.c:464 launch_asyncagi: No frame read > on channel SIP/501-08279028, going out ... > [Apr 13 12:03:57] DEBUG[2755]: pbx.c:2427 __ast_pbx_run: Spawn extension > (sip_sercom,500,0) exited non-zero on 'SIP/501-08279028' > [Apr 13 12:03:57] == Spawn extension (sip_sercom, 500, 0) exited non-zero > on 'SIP/501-08279028' > > By the way, there's another thing puzzling me: Due you said this AsyncAGI > patch was done for asterisk 1.6 and not for asterisk 1.4, and Henrik > Westerbeg said it had worked for it as well, (please see: > http://lists.digium.com/pipermail/asterisk-users/2008-December/223009.html) > then I looked over the last releases at > http://bugs.digium.com/bug_view_advanced_page.php?bug_id=11282 for that > AsyncAGI patch and I was able to see neither of them have the "returnstatus = > AGI_RESULT_HANGUP" either, however, ¡they work! (as Henrik said). > > As you can see, I'm a bit confusing about this subject. I would thank you If > you can give any guidelines about it in order to be able to investigate > deeper and move forward. > > Thank you very much for your help > Jose M Arias I really think you did not recompile and reinstall after applying the new patch. I don't see any code path where the message [Apr 13 12:03:57] DEBUG[2755]: res_agi.c:464 launch_asyncagi: No frame read on channel SIP/501-08279028, going out ... Is displayed but then ast_log(LOG_DEBUG, "launch_asyncagi returned (0x%X) for chan %s\n", returnstatus, chan->name); is NOT displayed. In fact, there is no way you can get out of launch_asyncagi without displaying that message. I tested this with 1.4.18 version exactly. The fact that works for some people and not for others may be due to different asterisk versions and/or dial plan specific issues. Please make sure the patch was correctly applied, once that is done we can try some other things. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ignoring time spent waiting in queue in CDR
Miguel Molina writes: [...] > Why don't you just make your billing statistics from the queue log? > Assuming that you upload the queue log to a database, it would be very easy. I took a look at this option and it looks like it will work. Thanks to all who responded for your advice! Scott. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RTCP ports
[Apr 15 11:12:19] ERROR[2624]: rtp.c:2447 ast_rtcp_write_sr: RTCP SR transmission error to aaa.bbb.ccc.ddd:37259, rtcp halted Operation not permitted [Apr 15 11:12:23] ERROR[2624]: rtp.c:2447 ast_rtcp_write_sr: RTCP SR transmission error to aaa.bbb.ccc.ddd:38563, rtcp halted Operation not permitted What is the specific nature of this traffic? Despite the above the call still functions. What is the appropriate FW rules? Michael ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 2B Channel Transfer on XO-based T1
Hi, I'm using it on 1.4. Actually, the driver does all the work. I have clients calling in, give them DISA, they dial the number and the call is pushed to the switch. Jimmy > -Original Message- > From: m...@povo.com > Sent: Tue, 14 Apr 2009 17:52:23 -0400 > To: asterisk-users@lists.digium.com > Subject: [asterisk-users] 2B Channel Transfer on XO-based T1 > > I'm trying to get "blind transfer" from an incoming DAHDI line to an > external number to work on an * 1.6 install using a T1 from XO. The > documentation is very "distributed" and incomplete, so while it's not > working, it's definitely more likely my error somehow. Couple questions > if > anybody is out there who even knows what TBCT is. > > > > 1) Is this even supported? > > 2) Does it require some settings in dahdi_channels, or features, or > whatever? > > 3) Would I "trigger" it via a Dial command or commands, or via > Transfer? > > 4) Do either or both of the legs need to be answered? > > > > Thanks very much. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MOH
Khaled W. Chehab wrote: > Any idead on how to begin with AGI > > That I don't. Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety." ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 2B Channel Transfer on XO-based T1
On Tue, 2009-04-14 at 17:52 -0400, Max Metral wrote: > I’m trying to get “blind transfer” from an incoming DAHDI line to an > external number to work on an * 1.6 install using a T1 from XO. The > documentation is very “distributed” and incomplete, so while it’s not > working, it’s definitely more likely my error somehow. Couple > questions if anybody is out there who even knows what TBCT is… > 1) Is this even supported? s Yes, it's supported in Asterisk and DAHDI, but your success in getting it to work will depend on many factors. As I understand it, it only works with certain switch types (I've had the best luck on 5ESS), and only when the telco enables that feature on your trunk group. In my experience, the telcos usually don't enable this feature by default, and it can be a pain to talk them into enabling it. > 2) Does it require some settings in dahdi_channels, or features, > or whatever? It requires the following features be enabled in chan_dahdi.conf (or zapata.conf, for later version of Zaptel): facilityenable=yes transfer=yes > > 3) Would I “trigger” it via a Dial command or commands, or via > Transfer? Neither... it happens automagically! Some time after the second leg of the call has answered, Asterisk will send a facility message to the CO switch saying "Hey, mind bridging these two calls on your end, so I can free up the channels on my end?" If the switch says "OK", you'll see the calls disappear from Asterisk (and the people on the calls won't know the difference). Otherwise, the calls will continue to be bridged by Asterisk. Obviously there are options to the Dial() application that would preclude Asterisk from allowing the transfer to happen, such as the t, T, w, and W options (and I'm sure there are probably more). > 4) Do either or both of the legs need to be answered? It's my understanding that both legs need to be answered and bridged before this will happen, but I'm not 100% sure. One other minor thing I'll point out... assuming that your 2-B-channel transfer is successful, the telco will send a message to Asterisk at the time the call is eventually hung up. Unfortunately, Asterisk has long since forgotten about the call by that point, so it simply writes a harmless warning message to the console and goes on its merry way. (If a developer happens to read this and needs a pet project -- it would be nice if this would update the CDR records for the original call!) I hope that's enough documentation to get you started! Please let us know how it works out for you! -- Jared Smith Training Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] duration of rfc2833 generated dtmf
Its not there and the link you gave me says its for sip originating rather than calls to a sip channel. on Tuesday 04/14/2009 Brent Davidson(br...@texascountrytitle.com) wrote > It's been around awhile. I've used it in 1.4 Check out this link for > basic info: http://www.voip-info.org/wiki/view/Asterisk+cmd+SIPdtmfmode > > John covici wrote: > > Thanks -- can not find sip dtmf mode or sip dtmfmode in asterisk-1.4. > > Is this new in 1.6? > > > > on Tuesday 04/14/2009 Brent Davidson(br...@texascountrytitle.com) wrote > > > To the best of my knowledge, the only way for you to control the > > > duration sent to the PSTN lines is for you to be directly connected to > > > the lines so you can set the tone duration in zapata.conf / dahdi.conf > > > or to use inband signalling. > > > > > > One thing you might try is researching the "SipDtmfMode" command. It > > > allows you to change the DTMF mode on an active channel. A suggestion > > > might be to set up the dial command with the M() option that point to a > > > Macro that changes the DTMF to INBAND once you are connected to the > > > problem number. At least in theory, if your provider is expecting > > > RFC2833 and they get inband, they should just ignore the inband > > > signaling and pass it on as part of the audio stream. The only problem > > > is that this may only work if you use uLaw or aLaw for your codec and I > > > don't know exactly how to set the tone duration without having a > > > zapata.conf or dahdi.conf entry. Even with one of those files, I don't > > > know how Asterisk chooses to do the rfc2833 to inband translation or > > > where it pulls the toneduration setting from if no PSTN interface is > > > involved in the call. > > > > > > -Brent > > > > > > John covici wrote: > > > > OK, thanks. If I could convince them to use info, would that be > > > > better as far as the duration is concerned? > > > > > > > > > > > > on Monday 04/13/2009 Brent Davidson(br...@texascountrytitle.com) wrote > > > > > John covici wrote: > > > > > > Hi. I have a SIP provider which wants RFC2833 for the dtmfmode, > > > > > > however I would like to increase the duration of the tone, its > > pretty > > > > > > short and some IVR's are unhappy or don't detect it. I did poke > > > > > > around, but it looks like when RFC2833 is used, it actually > > generates > > > > > > rtp packets of some sort, so I have no idea how to increase that > > > > > > duration. > > > > > > > > > > > > Any assistance would be appreciated. > > > > > > > > > > > > > > > > > > > > > > If your provider insists on rfc2833, then their servers will be > > > > > responsible for setting the tone duration sent to PSTN lines. > > > > > > > > > > > > > > > > > -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 2B Channel Transfer on XO-based T1
On 4/14/09, Max Metral wrote: > > I’m trying to get “blind transfer” from an incoming DAHDI line to an > external number to work on an * 1.6 install using a T1 from XO. The > documentation is very “distributed” and incomplete, so while it’s not > working, it’s definitely more likely my error somehow. Couple questions if > anybody is out there who even knows what TBCT is… > > > > 1) Is this even supported? > > 2) Does it require some settings in dahdi_channels, or features, or > whatever? > > 3) Would I “trigger” it via a Dial command or commands, or via > Transfer? > > 4) Do either or both of the legs need to be answered? Are you positive that your carrier PRI circuit has this feature enabled? If so, how much are they charging you for this service?(if they are not charging you for it monthly it is most likely not enabled) What kind of PRI do you have? (5ESS, NI2, DMS100,...) How many PRIs and trunk groups set up across them do you have? I have set up 2BCT for two different call center clients before, and neither implementation went smoothly as far as the carrier's part of it was concerned. Asterisk and zaptel(Dahdi) can handle it and will always attempt to do 2BCT on ALL native bridging of channels on the same trunk group if you have Transfer=yes in zapata.conf(or the Dahdi equivelent file). I have never configured 2BCT on an Asterisk 1.6 system, only 1.2 and 1.4 (using zaptel 1.4 for both), although I can't see any reason why it wouldn't work. MATT--- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 2B Channel Transfer on XO-based T1
I know what TBCT is, but haven't used it with Asterisk yet. Hopefully will be soon. 1) Supposed to be supported by Asterisk. Is supported by XO (I use it with a different platform). 2) Dunno 3) Dunno 4) The ISDN spec sez that one of the calls needs to be active to initiate the transfer. --Don Don Kelly PCF Corp People Come First 651 842-1000 888 Don Kell(y) 651 842-1001 fax _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Max Metral Sent: Tuesday, April 14, 2009 4:52 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] 2B Channel Transfer on XO-based T1 I'm trying to get "blind transfer" from an incoming DAHDI line to an external number to work on an * 1.6 install using a T1 from XO. The documentation is very "distributed" and incomplete, so while it's not working, it's definitely more likely my error somehow. Couple questions if anybody is out there who even knows what TBCT is. 1) Is this even supported? 2) Does it require some settings in dahdi_channels, or features, or whatever? 3) Would I "trigger" it via a Dial command or commands, or via Transfer? 4) Do either or both of the legs need to be answered? Thanks very much. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Exit Dial Application
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi,
Atis Lezdins schrieb:
>
> Ok, at first glance the app_macro looks suspicious, can You try
> calling dial without Macro?
Tried it without macro -> same behavior.
>
> If unsuccessful, You could enable debug level 2, it will tell way much
> more of everything, including DTMF events etc. Btw, does DTMF work at
> all for this Zap/ line? You could verify that by using Read before
> Dial.
Called Read, entered numbers, echoed them correctly.
Then I tried something ... different. I Answered the call before calling the
macro. And voila it's working. Do I have to answer the channel before Dial
option 'd' is working? It's a bit odd, cause the dial duration starts counting
and I hear a
'beep'. That's not ideal : / I've attached a full.log.
>
> Regards,
> Atis
>
chris...
- --
commpany dialog solutions gmbh
Dipl.-Ing.(FH) Christoph Fürstaller
IP-Communications
Ischlerbahnstraße 14, 5301 Eugendorf
Tel: +43 662 879512 Fax: +43 662 875960
IP-Tel: +43 780 commpany (26667269)
Email: c.fuerstal...@commpany.at
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.9 (MingW32)
iEYEARECAAYFAknlBuMACgkQR0exH8dhr/YSYwCeOcCfSlsnQIRff3L/F5wUvHh+
wCIAnRMC+YR7n7ZGmAvPKYbwZ7V/vc0O
=7cnt
-END PGP SIGNATURE-
[Apr 15 00:02:34] DEBUG[8816] pbx.c: Launching 'Answer'
[Apr 15 00:02:34] VERBOSE[8816] logger.c: -- Executing
[s-d...@macro-dialone:10] Answer("Zap/31-1", "") in new stack
[Apr 15 00:02:34] VERBOSE[8816] logger.c: -- Executing
[s-d...@macro-dialone:11] Set("Zap/31-1", "_EXITCONTEXT=callback") in new stack
[Apr 15 00:02:34] DEBUG[8816] app_macro.c: Executed application: Set
[Apr 15 00:02:34] DEBUG[8816] pbx.c: Launching 'Set'
[Apr 15 00:02:34] VERBOSE[8816] logger.c: -- Executing
[s-d...@macro-dialone:12] Set("Zap/31-1", "orig_exten=236") in new stack
[Apr 15 00:02:34] DEBUG[8816] app_macro.c: Executed application: Set
[Apr 15 00:02:34] DEBUG[8816] pbx.c: Launching 'Dial'
[Apr 15 00:02:34] VERBOSE[8816] logger.c: -- Executing
[s-d...@macro-dialone:13] Dial("Zap/31-1", "SIP/236|30|dtT") in new stack
[Apr 15 00:02:34] DEBUG[8816] chan_sip.c: Asked to create a SIP channel with
formats: 0x8 (alaw)
[Apr 15 00:02:34] DEBUG[8816] chan_sip.c: Allocating new SIP dialog for (No
Call-ID) - INVITE (With RTP)
[Apr 15 00:02:34] DEBUG[8816] chan_sip.c: Setting NAT on RTP to On
[Apr 15 00:02:34] DEBUG[8816] acl.c: # Testing 10.10.5.1 with 10.10.0.0
[Apr 15 00:02:34] DEBUG[8816] rtp.c: Channel 'Zap/31-1' has no RTP, not doing
anything
[Apr 15 00:02:34] DEBUG[8816] channel.c: Not copying variable MACRO_DEPTH.
[Apr 15 00:02:34] DEBUG[8816] channel.c: Not copying variable orig_exten.
[Apr 15 00:02:34] DEBUG[8816] channel.c: Copying soft-transferable variable
EXITCONTEXT.
[Apr 15 00:02:34] DEBUG[8816] channel.c: Not copying variable calls.
[Apr 15 00:02:34] DEBUG[8816] channel.c: Not copying variable peer.
[Apr 15 00:02:34] DEBUG[8816] channel.c: Not copying variable ARG2.
[Apr 15 00:02:34] DEBUG[8816] channel.c: Not copying variable DIALNUM.
[Apr 15 00:02:34] DEBUG[8816] channel.c: Not copying variable CFNA.
[Apr 15 00:02:34] DEBUG[8816] channel.c: Not copying variable CFBS.
[Apr 15 00:02:34] DEBUG[8816] channel.c: Not copying variable CFIM.
[Apr 15 00:02:34] DEBUG[8816] channel.c: Not copying variable COUNT.
[Apr 15 00:02:34] DEBUG[8816] channel.c: Not copying variable ARG1.
[Apr 15 00:02:34] DEBUG[8816] channel.c: Not copying variable MACRO_PRIORITY.
[Apr 15 00:02:34] DEBUG[8816] channel.c: Not copying variable MACRO_CONTEXT.
[Apr 15 00:02:34] DEBUG[8816] channel.c: Not copying variable MACRO_EXTEN.
[Apr 15 00:02:34] DEBUG[8816] channel.c: Copying soft-transferable variable
start.
[Apr 15 00:02:34] DEBUG[8816] channel.c: Copying soft-transferable variable
intern.
[Apr 15 00:02:34] DEBUG[8816] channel.c: Not copying variable CALLEDTON.
[Apr 15 00:02:34] DEBUG[8816] channel.c: Not copying variable ANI2.
[Apr 15 00:02:34] DEBUG[8816] channel.c: Not copying variable
TRANSFERCAPABILITY.
[Apr 15 00:02:34] DEBUG[8816] chan_sip.c: Outgoing Call for 236
[Apr 15 00:02:34] DEBUG[8816] chan_sip.c: Call to peer '236' is 1 out of 10
[Apr 15 00:02:34] DEBUG[8816] chan_sip.c: Our T38 capability (0), joint T38
capability (0)
[Apr 15 00:02:34] DEBUG[8816] chan_sip.c: ** Our capability: 0x10e
(gsm|ulaw|alaw|g729) Video flag: False
[Apr 15 00:02:34] DEBUG[8816] chan_sip.c: ** Our prefcodec: 0x8 (alaw)
[Apr 15 00:02:34] VERBOSE[8816] logger.c: -- Called 236
[Apr 15 00:02:34] DEBUG[8816] channel.c: Set channel SIP/236-081df8b0 to read
format slin
[Apr 15 00:02:34] DEBUG[8816] channel.c: Set channel Zap/31-1 to write format
slin
[Apr 15 00:02:34] DEBUG[8816] channel.c: Set channel Zap/31-1 to read format
g729
[Apr 15 00:02:34] DEBUG[7616] chan_sip.c: (Provisional) Stopping retransmission
(but retaining packet) on '45c32fd1024523451dba563865cd0...@xxx.at' Request
102: Found
[Apr 15 00:02:34] VERBOSE[8816] logger.c: -- SIP/236-081df8b0 is ringing
[Apr 15 00:02:34] DEB
[asterisk-users] 2B Channel Transfer on XO-based T1
I'm trying to get "blind transfer" from an incoming DAHDI line to an external number to work on an * 1.6 install using a T1 from XO. The documentation is very "distributed" and incomplete, so while it's not working, it's definitely more likely my error somehow. Couple questions if anybody is out there who even knows what TBCT is. 1) Is this even supported? 2) Does it require some settings in dahdi_channels, or features, or whatever? 3) Would I "trigger" it via a Dial command or commands, or via Transfer? 4) Do either or both of the legs need to be answered? Thanks very much. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] {Spam?} Vacation reply
On Tue, 2009-04-14 at 09:45 -0700, awerf...@hotmail.com wrote: > Hi Friend, > > How are you doing recently? I'm getting bored. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Exit Dial Application
On Tue, Apr 14, 2009 at 11:11 PM, Christoph Fürstaller wrote: > Thanks for the hint. I've looked aht the full log. I've attached a snipplet > from the file. But I can't see anythin which can help me. Very interesting, > but not helpful for me : / Is it possible to deactivate the 'd' option? Or > what else could cause > my problem? > Ok, at first glance the app_macro looks suspicious, can You try calling dial without Macro? If unsuccessful, You could enable debug level 2, it will tell way much more of everything, including DTMF events etc. Btw, does DTMF work at all for this Zap/ line? You could verify that by using Read before Dial. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs Inc, a...@iq-labs.net Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] duration of rfc2833 generated dtmf
It's been around awhile. I've used it in 1.4 Check out this link for basic info: http://www.voip-info.org/wiki/view/Asterisk+cmd+SIPdtmfmode John covici wrote: > Thanks -- can not find sip dtmf mode or sip dtmfmode in asterisk-1.4. > Is this new in 1.6? > > on Tuesday 04/14/2009 Brent Davidson(br...@texascountrytitle.com) wrote > > To the best of my knowledge, the only way for you to control the > > duration sent to the PSTN lines is for you to be directly connected to > > the lines so you can set the tone duration in zapata.conf / dahdi.conf > > or to use inband signalling. > > > > One thing you might try is researching the "SipDtmfMode" command. It > > allows you to change the DTMF mode on an active channel. A suggestion > > might be to set up the dial command with the M() option that point to a > > Macro that changes the DTMF to INBAND once you are connected to the > > problem number. At least in theory, if your provider is expecting > > RFC2833 and they get inband, they should just ignore the inband > > signaling and pass it on as part of the audio stream. The only problem > > is that this may only work if you use uLaw or aLaw for your codec and I > > don't know exactly how to set the tone duration without having a > > zapata.conf or dahdi.conf entry. Even with one of those files, I don't > > know how Asterisk chooses to do the rfc2833 to inband translation or > > where it pulls the toneduration setting from if no PSTN interface is > > involved in the call. > > > > -Brent > > > > John covici wrote: > > > OK, thanks. If I could convince them to use info, would that be > > > better as far as the duration is concerned? > > > > > > > > > on Monday 04/13/2009 Brent Davidson(br...@texascountrytitle.com) wrote > > > > John covici wrote: > > > > > Hi. I have a SIP provider which wants RFC2833 for the dtmfmode, > > > > > however I would like to increase the duration of the tone, its > pretty > > > > > short and some IVR's are unhappy or don't detect it. I did poke > > > > > around, but it looks like when RFC2833 is used, it actually > generates > > > > > rtp packets of some sort, so I have no idea how to increase that > > > > > duration. > > > > > > > > > > Any assistance would be appreciated. > > > > > > > > > > > > > > > > > > If your provider insists on rfc2833, then their servers will be > > > > responsible for setting the tone duration sent to PSTN lines. > > > > > > > > > > > > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What means? Correct auth, but based on stale nonce received
Y, it can be that someone wants to register with a sniffed SIP packet. it's basically the nonce="" value is not the same Asterisk sent for that REGISTER session Martin On Tue, Apr 14, 2009 at 11:10 AM, Danny Nicholas wrote: > http://lists.digium.com/pipermail/asterisk-users/2005-July/110220.html > > It means that a SIP device is re-using an old authentication challenge. > If it still registers and can place calls, there's no problem to worry > about. It's just a warning. > > > -Original Message- > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tiago Durante > Sent: Tuesday, April 14, 2009 11:06 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [asterisk-users] What means? Correct auth,but based on stale nonce > received > > Hi masters! > > I've this Asterisk 1.4.15 running. yesterday I had to change the > firewall schema that I had before. > > I use to have a FW that would be my network FW/Proxy and do the NATs > for Asterisk. This FW was receiving too many requests from my LAN and > it was making the Asterisk 'cut' the calls or reach very high latency. > > Yesterday I added a new FW just for this Asterisk. The same > configuration as the old firewall, loading the same modules, same > NATs. > > But now some ATAs (sip) can't register against this Asterisk and the > ones that can generates these messages: > > [Apr 14 13:01:45] NOTICE[29235]: chan_sip.c:8375 check_auth: Correct > auth, but based on stale nonce received from > 'sip:xx...@200.x.x.x:5060' > [Apr 14 13:01:45] NOTICE[29235]: chan_sip.c:8375 check_auth: Correct > auth, but based on stale nonce received from > 'sip:xx...@200.x.x.x:5060' > > Does any one know why this happens? > > Thank you!!! > > -- > Tiago Durante > > ,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,., > Perseverance is the hard work you do after you > get tired of doing the hard work you already did. > -- Newt Gingrich > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Exit Dial Application
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi,
Atis Lezdins schrieb:
> That's CLI interface output, log should have timestamps and much more
> detail in it.
>
> Check /var/log/asterisk/full (assuming default install location).
> You'll need to enable "full" line in logger.conf, restart Asterisk and
> issue "core set verbose 3" and "core set debug 1" in CLI.
Thanks for the hint. I've looked aht the full log. I've attached a snipplet
from the file. But I can't see anythin which can help me. Very interesting, but
not helpful for me : / Is it possible to deactivate the 'd' option? Or what
else could cause
my problem?
>
>
> Regards,
> Atis
thanks for your help,
chris...
- --
commpany dialog solutions gmbh
Dipl.-Ing.(FH) Christoph Fürstaller
IP-Communications
Ischlerbahnstraße 14, 5301 Eugendorf
Tel: +43 662 879512 Fax: +43 662 875960
IP-Tel: +43 780 commpany (26667269)
Email: c.fuerstal...@commpany.at
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.9 (MingW32)
iEYEARECAAYFAknk7gMACgkQR0exH8dhr/Y+1QCfTM8FvjA/9Zim7m9QbdjTYbQc
QGQAnR92l1smtrs8Ao8f0vlaEdHiQv3R
=KE+7
-END PGP SIGNATURE-
[Apr 14 22:49:25] VERBOSE[7867] logger.c: -- Executing
[s-d...@macro-dialone:11] Set("Zap/31-1", "EXITCONTEXT=callback") in new stack
[Apr 14 22:49:25] DEBUG[7867] app_macro.c: Executed application: Set
[Apr 14 22:49:25] DEBUG[7867] pbx.c: Launching 'Set'
[Apr 14 22:49:25] VERBOSE[7867] logger.c: -- Executing
[s-d...@macro-dialone:12] Set("Zap/31-1", "orig_exten=236") in new stack
[Apr 14 22:49:25] DEBUG[7867] app_macro.c: Executed application: Set
[Apr 14 22:49:25] DEBUG[7867] pbx.c: Launching 'Dial'
[Apr 14 22:49:25] VERBOSE[7867] logger.c: -- Executing
[s-d...@macro-dialone:13] Dial("Zap/31-1", "SIP/236|30|dtT") in new stack
[Apr 14 22:49:25] DEBUG[7867] chan_sip.c: Asked to create a SIP channel with
formats: 0x8 (alaw)
[Apr 14 22:49:25] DEBUG[7867] chan_sip.c: Allocating new SIP dialog for (No
Call-ID) - INVITE (With RTP)
[Apr 14 22:49:25] DEBUG[7867] chan_sip.c: Setting NAT on RTP to On
[Apr 14 22:49:25] DEBUG[7867] acl.c: # Testing 10.10.5.1 with 10.10.0.0
[Apr 14 22:49:25] DEBUG[7867] rtp.c: Channel 'Zap/31-1' has no RTP, not doing
anything
[Apr 14 22:49:25] DEBUG[7867] channel.c: Not copying variable MACRO_DEPTH.
[Apr 14 22:49:25] DEBUG[7867] channel.c: Not copying variable orig_exten.
[Apr 14 22:49:25] DEBUG[7867] channel.c: Not copying variable EXITCONTEXT.
[Apr 14 22:49:25] DEBUG[7867] channel.c: Not copying variable calls.
[Apr 14 22:49:25] DEBUG[7867] channel.c: Not copying variable peer.
[Apr 14 22:49:25] DEBUG[7867] channel.c: Not copying variable ARG2.
[Apr 14 22:49:25] DEBUG[7867] channel.c: Not copying variable DIALNUM.
[Apr 14 22:49:25] DEBUG[7867] channel.c: Not copying variable CFNA.
[Apr 14 22:49:25] DEBUG[7867] channel.c: Not copying variable CFBS.
[Apr 14 22:49:25] DEBUG[7867] channel.c: Not copying variable CFIM.
[Apr 14 22:49:25] DEBUG[7867] channel.c: Not copying variable COUNT.
[Apr 14 22:49:25] DEBUG[7867] channel.c: Not copying variable ARG1.
[Apr 14 22:49:25] DEBUG[7867] channel.c: Not copying variable MACRO_PRIORITY.
[Apr 14 22:49:25] DEBUG[7867] channel.c: Not copying variable MACRO_CONTEXT.
[Apr 14 22:49:25] DEBUG[7867] channel.c: Not copying variable MACRO_EXTEN.
[Apr 14 22:49:25] DEBUG[7867] channel.c: Copying soft-transferable variable
start.
[Apr 14 22:49:25] DEBUG[7867] channel.c: Copying soft-transferable variable
intern.
[Apr 14 22:49:25] DEBUG[7867] channel.c: Not copying variable CALLEDTON.
[Apr 14 22:49:25] DEBUG[7867] channel.c: Not copying variable ANI2.
[Apr 14 22:49:25] DEBUG[7867] channel.c: Not copying variable
TRANSFERCAPABILITY.
[Apr 14 22:49:25] DEBUG[7867] chan_sip.c: Outgoing Call for 236
[Apr 14 22:49:25] VERBOSE[7867] logger.c: -- Called 236
[Apr 14 22:49:25] DEBUG[7867] channel.c: Set channel SIP/236-08219bb0 to read
format slin
[Apr 14 22:49:25] DEBUG[7867] channel.c: Set channel Zap/31-1 to write format
slin
[Apr 14 22:49:25] DEBUG[7867] channel.c: Set channel Zap/31-1 to read format
g729
[Apr 14 22:49:25] DEBUG[7616] chan_sip.c: (Provisional) Stopping retransmission
(but retaining packet) on '65fc078c0c6cd24f5c068b770dabc...@xxx.at' Request
102: Found
[Apr 14 22:49:25] VERBOSE[7867] logger.c: -- SIP/236-08219bb0 is ringing
[Apr 14 22:49:25] DEBUG[7867] rtp.c: Channel 'Zap/31-1' has no RTP, not doing
anything
[Apr 14 22:49:25] DEBUG[7867] chan_zap.c: Requested indication 3 on channel
Zap/31-1
[Apr 14 22:49:26] DEBUG[7616] chan_sip.c: (Provisional) Stopping retransmission
(but retaining packet) on '65fc078c0c6cd24f5c068b770dabc...@xxx.at' Request
102: Found
[Apr 14 22:49:26] VERBOSE[7867] logger.c: -- SIP/236-08219bb0 is ringing
[Apr 14 22:49:26] DEBUG[7867] rtp.c: Channel 'Zap/31-1' has no RTP, not doing
anything
[Apr 14 22:49:27] DEBUG[7616] chan_sip.c: (Provisional) Stopping retransmission
(but retaining packet) on '65fc078c0c6cd24f5c068b770dabc...@xxx.at' Request
102: Found
[Apr 14 22:49:27] VERB
Re: [asterisk-users] Exit Dial Application
On Tue, Apr 14, 2009 at 9:14 PM, Christoph Fürstaller
wrote:
> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
>
> Hi Atis,
>
> No problem : ) I tried it again, here is the log output:
> -- Executing [...@from-pbx:1] Set("Zap/31-1", "EXITCONTEXT=callback") in
> new stack
> -- Executing [...@from-pbx:2] Dial("Zap/31-1", "SIP/236||d") in new stack
> -- Called 236
> -- SIP/236-0825f928 is ringing
> -- SIP/236-0825f928 is ringing
> -- SIP/236-0825f928 is ringing
> -- SIP/236-0825f928 is ringing
That's CLI interface output, log should have timestamps and much more
detail in it.
Check /var/log/asterisk/full (assuming default install location).
You'll need to enable "full" line in logger.conf, restart Asterisk and
issue "core set verbose 3" and "core set debug 1" in CLI.
Regards,
Atis
--
Atis Lezdins,
VoIP Project Manager / Developer,
IQ Labs Inc,
a...@iq-labs.net
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MOH
Any idead on how to begin with AGI -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle Sent: Tuesday, April 14, 2009 10:29 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] MOH Khaled W. Chehab wrote: > Man :) > I want the MOH play until Asterisk receives 180 ringing or 183 from the > termination GW. > I don't think you'll be able to mix and match via the dial application. You may have to try using AGI for this. That, I can't help you with. Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety." ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] duration of rfc2833 generated dtmf
Thanks -- can not find sip dtmf mode or sip dtmfmode in asterisk-1.4. Is this new in 1.6? on Tuesday 04/14/2009 Brent Davidson(br...@texascountrytitle.com) wrote > To the best of my knowledge, the only way for you to control the > duration sent to the PSTN lines is for you to be directly connected to > the lines so you can set the tone duration in zapata.conf / dahdi.conf > or to use inband signalling. > > One thing you might try is researching the "SipDtmfMode" command. It > allows you to change the DTMF mode on an active channel. A suggestion > might be to set up the dial command with the M() option that point to a > Macro that changes the DTMF to INBAND once you are connected to the > problem number. At least in theory, if your provider is expecting > RFC2833 and they get inband, they should just ignore the inband > signaling and pass it on as part of the audio stream. The only problem > is that this may only work if you use uLaw or aLaw for your codec and I > don't know exactly how to set the tone duration without having a > zapata.conf or dahdi.conf entry. Even with one of those files, I don't > know how Asterisk chooses to do the rfc2833 to inband translation or > where it pulls the toneduration setting from if no PSTN interface is > involved in the call. > > -Brent > > John covici wrote: > > OK, thanks. If I could convince them to use info, would that be > > better as far as the duration is concerned? > > > > > > on Monday 04/13/2009 Brent Davidson(br...@texascountrytitle.com) wrote > > > John covici wrote: > > > > Hi. I have a SIP provider which wants RFC2833 for the dtmfmode, > > > > however I would like to increase the duration of the tone, its pretty > > > > short and some IVR's are unhappy or don't detect it. I did poke > > > > around, but it looks like when RFC2833 is used, it actually generates > > > > rtp packets of some sort, so I have no idea how to increase that > > > > duration. > > > > > > > > Any assistance would be appreciated. > > > > > > > > > > > > > > If your provider insists on rfc2833, then their servers will be > > > responsible for setting the tone duration sent to PSTN lines. > > > > > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MOH
Khaled W. Chehab wrote: > Man :) > I want the MOH play until Asterisk receives 180 ringing or 183 from the > termination GW. > I don't think you'll be able to mix and match via the dial application. You may have to try using AGI for this. That, I can't help you with. Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety." ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MOH
Man :)
I want the MOH play until Asterisk receives 180 ringing or 183 from the
termination GW.
Here I want to stop the MOH and let the user hear the early media RBT
Regards
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle
Sent: Tuesday, April 14, 2009 9:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] MOH
Khaled W. Chehab wrote:
> Thanks for answering Doug
>
>
> I am using exten => _X.,n,Dial(SIP/OPNS/${EXTEN}|300|m) with no macros
>
Change this to:
_X.,n,Dial(SIP/OPNS/${EXTEN}|300)
The m was causing the music on hold.
Doug
--
Ben Franklin quote:
"Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety."
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
*
No employee or agent is authorized to conclude any binding agreement on behalf
of Xplorium with another party by e-mail without express written confirmation
by an officer of Xplorium. Any views expressed by an individual in this
electronic message do not necessarily reflect views of Xplorium or its
subsidiaries and associates.
This electronic message and its attachments are solely addressed to the
addressee(s), and contain confidential information protected from disclosure
belonging to Xplorium.
If you are not the intended addressee of this electronic message and its
attachments, kindly delete it immediately from your system and notify the
sender by electronic mail. You must not copy this message or attachment or
disclose its content to any other person.
Xplorium does not guarantee the integrity of this electronic message and any of
its attachments, or that they are free from computer viruses or other defects.
*
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Vacation reply user nuked
John Todd wrote: > Sorry for the brief interruption. The user spamming the list with > vacation replies has been removed. > > Bless you! Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety." ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MOH
Khaled W. Chehab wrote:
> Thanks for answering Doug
>
>
> I am using exten => _X.,n,Dial(SIP/OPNS/${EXTEN}|300|m) with no macros
>
Change this to:
_X.,n,Dial(SIP/OPNS/${EXTEN}|300)
The m was causing the music on hold.
Doug
--
Ben Franklin quote:
"Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety."
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dial a pager and enter DTMF
This worked: Dial(SIP/5198881...@telasip-gw4,,D(12345678)) > > however, the problem now exists in the disconnection. Asterisk tries to > bridge the call, play dtmf but never disconnects. What is there a specific > syntax to the D command that specifies a disconnect period. > I am thinking a better solution might just be adding it to a .call file > and running a script when I need to page my doctors. That way I don't have > to stay on the line and listen to the series of calls. On Mon, Feb 26, 2007 at 1:43 AM, Yuan LIU wrote: > From: Supa >> Date: Sun, 25 Feb 2007 15:45:08 -0500 >> >> this dials, and upon answers plays dtmf tones, but does not auto >> disconnect: >> >> exten => s,2,Dial(SIP/TelaSip-gw4/5198881212,15,D(12345678),S(8)) >> >> and this disconnects after 8 secs, but does not play dtmf: >> >> exten => s,2,Dial(SIP/TelaSip-gw4/5198881212,15,S(8),D(12345678)) >> > > Thought it would be (note the syntax: no commas between flags) > > Dial(SIP/5198881...@telasip-gw4,15,D(12345678)L(8000)) > > Is S() a new flag? > > Yuan Liu > > any ideas of what wrong with the above syntax >> >> On 2/25/07, Supa wrote: >> >>> >>> This worked: >>> Dial(SIP/5198881...@telasip-gw4,,D(12345678)) >>> >>> however, the problem now exists in the disconnection. Asterisk tries to >>> bridge the call, play dtmf but never disconnects. What is there a >>> specific >>> syntax to the D command that specifies a disconnect period. >>> I am thinking a better solution might just be adding it to a .call file >>> and running a script when I need to page my doctors. That way I don't >>> have >>> to stay on the line and listen to the series of calls. >>> >> > > ___ > --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 http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk-beginner : cannot make phonecalls using Asterisk
On Tuesday 14 April 2009 13:04:02 jonas kellens wrote: > -A RH-Firewall-1-INPUT -j REJECT --reject-with icmp-host-prohibited > -A RH-Firewall-1-INPUT -p udp -m udp --dport 5060 -j ACCEPT For one, these two rules are reversed. For two, you've failed to create holes in your firewall for UDP/1-2 or whatever range you're using for RTP. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk-beginner : cannot make phonecalls using Asterisk
Your problem is that you put your line after the REJECT line. Your line is never reached. Move it up one line, before the REJECT, and it will work as expected. // T _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of jonas kellens Sent: den 14 april 2009 20:04 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk-beginner : cannot make phonecalls using Asterisk There is something wrong with my IPtables !!! When i do : service iptables stop I see my phones register on the CLI !! I can place a call and the phone rings !! I see a whole lot of SIP-requests on the CLI with SDP-message in body !! That's good news... What is wrong with my IPtables-rule I've added in /etc/sysconfig/iptables ??? [r...@asterisk sysconfig]# cat iptables # Firewall configuration written by system-config-securitylevel # Manual customization of this file is not recommended. *filter :INPUT ACCEPT [0:0] :FORWARD ACCEPT [0:0] :OUTPUT ACCEPT [0:0] :RH-Firewall-1-INPUT - [0:0] -A INPUT -j RH-Firewall-1-INPUT -A FORWARD -j RH-Firewall-1-INPUT -A RH-Firewall-1-INPUT -i lo -j ACCEPT -A RH-Firewall-1-INPUT -p icmp --icmp-type any -j ACCEPT -A RH-Firewall-1-INPUT -p 50 -j ACCEPT -A RH-Firewall-1-INPUT -p 51 -j ACCEPT -A RH-Firewall-1-INPUT -p udp --dport 5353 -d 224.0.0.251 -j ACCEPT -A RH-Firewall-1-INPUT -p udp -m udp --dport 631 -j ACCEPT -A RH-Firewall-1-INPUT -p tcp -m tcp --dport 631 -j ACCEPT -A RH-Firewall-1-INPUT -m state --state ESTABLISHED,RELATED -j ACCEPT -A RH-Firewall-1-INPUT -m state --state NEW -m tcp -p tcp --dport 22 -j ACCEPT -A RH-Firewall-1-INPUT -j REJECT --reject-with icmp-host-prohibited -A RH-Firewall-1-INPUT -p udp -m udp --dport 5060 -j ACCEPT COMMIT Greetingz, Jonas. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Gxp 2000 softkey question
I have a function *1 that starts and stops recording in a call. I use a function so I can use MixMonitor. It works well, however I would like to make it a little more integrated for my users. We have GXP 200 hardphones. So far I've been able to configure a softkey using the speeddial option to dial *1 during a call. I also have setup another key to monitor the status of recording (on/off) using the BLF function and the devstate function. (new in 1.6 back ported to 1.4) However I seem to be unable to combine these functions in to a single key. Can anyone offer any assistance? Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 da...@safedatausa.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Exit Dial Application
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi Atis,
No problem : ) I tried it again, here is the log output:
-- Executing [...@from-pbx:1] Set("Zap/31-1", "EXITCONTEXT=callback") in
new stack
-- Executing [...@from-pbx:2] Dial("Zap/31-1", "SIP/236||d") in new stack
-- Called 236
-- SIP/236-0825f928 is ringing
-- SIP/236-0825f928 is ringing
-- SIP/236-0825f928 is ringing
-- SIP/236-0825f928 is ringing
Nothing happens. I adopted my [callback] context:
[callback]
exten => 1,1,Verbose(hello)
exten => s,1,Verbose(s)
exten => i,1,Verbose(i)
exten => 5,1,agi(str_concat.sh)
exten => 5,n,Hangup
But nothing happens, if I dial 1, 5, or everything else. I have no clue what's
wrong here.
chris...
Atis Lezdins schrieb:
>> Thanks for your replay. But in my 1st post, I mentioned my dial statement:
>> exten => _X.,n,Dial(${DIALNUM},${ARG2},dtT)
>>
>> As you can see, there is a d to exit the dial application. And one priority
>> earlier, I set the EXITCONTEXT variable. So everything _should_ work, but it
>> doesn't : /
>>
>
> Oh, sorry, missed that part :)
>
> Try enabling "full" log in logger.conf, set verbosity to 3 and debug
> to 1, and see what goes in it.
>
> Regards,
> Atis
>
- --
commpany dialog solutions gmbh
Dipl.-Ing.(FH) Christoph Fürstaller
IP-Communications
Ischlerbahnstraße 14, 5301 Eugendorf
Tel: +43 662 879512 Fax: +43 662 875960
IP-Tel: +43 780 commpany (26667269)
Email: c.fuerstal...@commpany.at
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.9 (MingW32)
iEYEARECAAYFAknk0qMACgkQR0exH8dhr/azTQCeIJqkCJxC/z5WHnIEoWcpgn8I
Xo4AoJf3DRn5zNqmUrME7hw4hBQluRM3
=7V9F
-END PGP SIGNATURE-
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Changing menuselect values from CLI and not TUI
On Tue, Apr 14, 2009 at 10:28:42AM +1000, David Klaverstyn wrote: > Hi All, > > I'm in the process of writing an install script and I would like to change > some settings for the install process but I don't want the user to go into > menuselect and make the changes manually. > > Is there a way to make the changes to menuselect from the CLI? > > As an example, selecting the iLBC codec. > menuselect codec ilbc on On the Debian package I generally patch source files and XML SPECs to set / reset the "defaltenable" options. The problem with patching the output of menuselect is that menuselect is run on each tim you un 'make'. This is why I never bothered adding such an option to dummy-select. OTOH, in dummy-select the configuration is straight-forward: echo enable codec_ilbc >build_tools/conf Menuselect uses the same file as both an input and an output, and this makes it very confusing and error-prone. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Dial Pagers And Enter Callback Numbers
Anyone have a asterisk pager script that dials a list of pager numbers, enters the phone number, and enters pound to send the page? I am looking for a perl or php solution. Ideally I would like to set the pager area-code and prefix on the fly as the company I work for owns a few blocks of pagers (111-111-11XX). I would like to specify the area code, prefix, and first two of the telephone number, pager call back number, and the number of calls to make. I would be willing to pay for a simple shell script ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MOH
Thanks for answering Doug
I am using exten => _X.,n,Dial(SIP/OPNS/${EXTEN}|300|m) with no macros
kindly can you wrote down a macro to stop the MOH RTP in order to let the
GW inband early media rtp heard by the caller
Regards
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle
Sent: Tuesday, April 14, 2009 8:57 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] MOH
Khaled W. Chehab wrote:
> Dear Ben,
>
> I tried a lot ,Kindly can you give me an example on how to do that using a
> macro.
>
> Remove the 'm' out of your dial command:
>
> Doug
>
>
I'm not Ben, but I'll answer.
Shows us what your macro looks like and we'll chime in with some pointers.
Doug
--
Ben Franklin quote:
"Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety."
Dears
-How can I stop MOH when status of the dial is ringing and let the user hear
the Ring Back Tone from the termination Gateway.
Even I can see in the CLI debugging the status is ringing -my idea is to add
music on hold stop when asterisk detect -- SIP/OPNS-096456c0 is ringing
line
In which script this line located?
-- Executing [97130245...@default:1]
SetMusicOnHold("SIP/xx.xx.xx.xx-096ca8c0", "English") in new stack
-- Executing [9713024...@default:2] Dial("SIP/xx.xx.xx.xx-096ca8c0",
"SIP/OPNS/9713024561|300|m") in new stack
-- Called OPNS/9713024561
-- Started music on hold, class 'English', on SIP/xx.xx.xx.xx-096ca8c0
-- SIP/OPNS-096456c0 is ringing
-More over when I used directrtpsetup=yes I heard the MOH and the ring back
tone together .
Regards
*
No employee or agent is authorized to conclude any binding agreement on behalf
of Xplorium with another party by e-mail without express written confirmation
by an officer of Xplorium. Any views expressed by an individual in this
electronic message do not necessarily reflect views of Xplorium or its
subsidiaries and associates.
This electronic message and its attachments are solely addressed to the
addressee(s), and contain confidential information protected from disclosure
belonging to Xplorium.
If you are not the intended addressee of this electronic message and its
attachments, kindly delete it immediately from your system and notify the
sender by electronic mail. You must not copy this message or attachment or
disclose its content to any other person.
Xplorium does not guarantee the integrity of this electronic message and any of
its attachments, or that they are free from computer viruses or other defects.
*
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Exit Dial Application
>
> Thanks for your replay. But in my 1st post, I mentioned my dial statement:
> exten => _X.,n,Dial(${DIALNUM},${ARG2},dtT)
>
> As you can see, there is a d to exit the dial application. And one priority
> earlier, I set the EXITCONTEXT variable. So everything _should_ work, but it
> doesn't : /
>
Oh, sorry, missed that part :)
Try enabling "full" log in logger.conf, set verbosity to 3 and debug
to 1, and see what goes in it.
Regards,
Atis
--
Atis Lezdins,
VoIP Project Manager / Developer,
IQ Labs Inc,
a...@iq-labs.net
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Vacation reply user nuked
Sorry for the brief interruption. The user spamming the list with vacation replies has been removed. JT --- John Todd email:jt...@digium.com Digium, Inc. | Asterisk Open Source Community Director 445 Jan Davis Drive NW - Huntsville AL 35806 - USA direct: +1-256-428-6083 http://www.digium.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] FW: Asterisk-beginner : cannot make phone calls using Asterisk
_ From: Cary Fitch [mailto:ca...@usawide.net] Sent: Tuesday, April 14, 2009 1:17 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Asterisk-beginner : cannot make phone calls using Asterisk May I suggest divide and conquer? I haven't followed every detail, but it seems that your phones are not registering. Put them on the same net as the sip server and get them to register. Then get it to where you can make a call from one to the other. Then back off through your router or what ever, with what ever filtering you have in place. In other words get it working "up close" then move out. until you break it, and find that problem. With at least one phone working, you can be sure the system is "good" at any instant and then make the other "distant" phone work too. I hope this helps. If I have missed the mark, explain more as to the point it is failing. Cary Fitch _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of jonas kellens Sent: Tuesday, April 14, 2009 12:58 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk-beginner : cannot make phonecalls using Asterisk I will summarize everything again and try to answer all the questions asked while I was away. First I stop Asterisk : [r...@asterisk asterisk]# /usr/sbin/asterisk -r Asterisk 1.4.24, Copyright (C) 1999 - 2008 Digium, Inc. and others. Created by Mark Spencer Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. = Connected to Asterisk 1.4.24 currently running on asterisk (pid = 3189) Verbosity is at least 3 asterisk*CLI> stop now asterisk*CLI> Disconnected from Asterisk server [r...@asterisk asterisk]# ps aux | grep asterisk avahi 3320 0.0 0.0 2588 1344 ?Ss 18:49 0:00 avahi-daemon: running [asterisk.local] root 3563 0.0 0.0 3912 676 pts/0S+ 19:11 0:00 grep asterisk ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Exit Dial Application
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi Atis,
Thanks for your replay. But in my 1st post, I mentioned my dial statement:
exten => _X.,n,Dial(${DIALNUM},${ARG2},dtT)
As you can see, there is a d to exit the dial application. And one priority
earlier, I set the EXITCONTEXT variable. So everything _should_ work, but it
doesn't : /
Chris...
Atis Lezdins schrieb:
> CLI> core show application Dial
>
> d- Allow the calling user to dial a 1 digit extension while waiting
> for
>a call to be answered. Exit to that extension if it exists in the
>current context, or the context defined in the EXITCONTEXT
> variable,
>if it exists.
>
> Regards,
> Atis
>
> On Tue, Apr 14, 2009 at 7:49 PM, Christoph Fürstaller
> wrote:
> Hi,
>
> Thanks for your replay. But this can only be done before or after the dial,
> but I wanna do it during the dial, when user A is waiting for user B,
> answering the phone. This should be possible, right?
>
> I hope anyone knows if this is possible.
>
> Chris...
>
> Danny Nicholas schrieb:
I'd change callback to this
[callback]
Exten => s,1,Playback(press5msg)
Exten => s,n,Waitexten(5)
Exten => s,n,Hangup
exten => 5,1,agi(str_concat.sh)
exten => 5,n,Hangup
This will play a message, wait 5 seconds for user to press 5, then hangup
if
they don't.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Christoph
Fuerstaller
Sent: Tuesday, April 14, 2009 5:04 AM
To: Asterisk Users Mailing List
Subject: [asterisk-users] Exit Dial Application
Hi,
I' try to implement an automatic callback mechanism, just for local SIP
calls.. Callback
on busy and on no answer. If the other party doen't answer, it should be
possible to press
5 to place an callback.
Here is my dial:
exten => _X.,1,Set(EXITCONTEXT=callback)
exten => _X.,n,Dial(${DIALNUM},${ARG2},dtT)
And here the script for callback.
[callback]
exten => 5,1,agi(str_concat.sh)
exten => 5,n,Hangup
If I call someone and press 5, nothing happens. What could be a problem?
DTMFmode is RFC2833 for all SIP Accounts. DTMF's are transmitted correctly,
I can enter
the voicmail menue.
I'm using Asterisk 1.4.21.1.
Any successions are very appreciated.
Chris...
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>>
___
- -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
>>
- --
commpany dialog solutions gmbh
Dipl.-Ing.(FH) Christoph Fürstaller
IP-Communications
Ischlerbahnstraße 14, 5301 Eugendorf
Tel: +43 662 879512 Fax: +43 662 875960
IP-Tel: +43 780 commpany (26667269)
Email: c.fuerstal...@commpany.at
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.9 (MingW32)
iEUEARECAAYFAknkzdAACgkQR0exH8dhr/YHPwCYgN8T2hBUEb/TrH95xh/WRcil
gwCgjvph3l5lcnJucuFURi2L8rySVD4=
=UJqh
-END PGP SIGNATURE-
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ring All Queue
On Tue, Apr 14, 2009 at 9:09 PM, Jim Dickenson wrote:
> At least in version 1.6.0.x you can specify a macro to be executed when the
> agent answers the queued call. This is an argument to the queue application.
>
> Queue(queuename[,options[,URL][,announceoverride][,timeout][,AGI][,macro][,g
> osub][,rule])
>
> The optional macro parameter will run a macro on the calling party's channel
> once they are connected to a queue member.
>
> Here is what my Macro does:
>
> exten => s,1,UserEvent(DidQueue,ActionID:${CfMC_ActionID} & ${UNIQUEID}
> & ${CHANNEL} & ${CfMC_AgentToUse} & ${CfMC_DialInfo} &
> ${CfMC_QueueToUse} & ${MEMBERINTERFACE} & ${MEMBERNAME})
>
> ${MEMBERINTERFACE} and ${MEMBERNAME} have info about the agent that answered
> the call.
Just test this with multiple simultenous answers, so You don't get any
surprises. I'd recommend putting Wait(10) into that macro (actually
GoSub in 1.6) and trying to pick up second ringing phone while first
is in Wait().
I haven't gotten into 1.6 yet, but here are some related problems on
1.4 with some backports:
http://bugs.digium.com/view.php?id=13335
http://bugs.digium.com/view.php?id=14859
Once You'll get the agent in some variable within answer part of
dialplan, it's just a matter of storing this into per-call database
entry and reading from parrent channel. See "function DB" and variable
"UNIQUEID" for that.
Of course, if You need it only on hangup, Luis suggestion will work
just fine, use Asterisk Realtime engine to read value from realtime
queue log.
Regards,
Atis
--
Atis Lezdins,
VoIP Project Manager / Developer,
IQ Labs Inc,
a...@iq-labs.net
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Exit Dial Application
CLI> core show application Dial
d- Allow the calling user to dial a 1 digit extension while waiting for
a call to be answered. Exit to that extension if it exists in the
current context, or the context defined in the EXITCONTEXT variable,
if it exists.
Regards,
Atis
On Tue, Apr 14, 2009 at 7:49 PM, Christoph Fürstaller
wrote:
> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
>
> Hi,
>
> Thanks for your replay. But this can only be done before or after the dial,
> but I wanna do it during the dial, when user A is waiting for user B,
> answering the phone. This should be possible, right?
>
> I hope anyone knows if this is possible.
>
> Chris...
>
> Danny Nicholas schrieb:
>> I'd change callback to this
>> [callback]
>> Exten => s,1,Playback(press5msg)
>> Exten => s,n,Waitexten(5)
>> Exten => s,n,Hangup
>> exten => 5,1,agi(str_concat.sh)
>> exten => 5,n,Hangup
>>
>> This will play a message, wait 5 seconds for user to press 5, then hangup if
>> they don't.
>>
>> -Original Message-
>> From: asterisk-users-boun...@lists.digium.com
>> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Christoph
>> Fuerstaller
>> Sent: Tuesday, April 14, 2009 5:04 AM
>> To: Asterisk Users Mailing List
>> Subject: [asterisk-users] Exit Dial Application
>>
>> Hi,
>>
>> I' try to implement an automatic callback mechanism, just for local SIP
>> calls.. Callback
>> on busy and on no answer. If the other party doen't answer, it should be
>> possible to press
>> 5 to place an callback.
>>
>> Here is my dial:
>> exten => _X.,1,Set(EXITCONTEXT=callback)
>> exten => _X.,n,Dial(${DIALNUM},${ARG2},dtT)
>>
>> And here the script for callback.
>> [callback]
>> exten => 5,1,agi(str_concat.sh)
>> exten => 5,n,Hangup
>>
>> If I call someone and press 5, nothing happens. What could be a problem?
>> DTMFmode is RFC2833 for all SIP Accounts. DTMF's are transmitted correctly,
>> I can enter
>> the voicmail menue.
>>
>> I'm using Asterisk 1.4.21.1.
>>
>> Any successions are very appreciated.
>>
>> Chris...
>
> ___
> - -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
> ___
> - -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
> - --
> commpany dialog solutions gmbh
>
> Dipl.-Ing.(FH) Christoph Fürstaller
> IP-Communications
>
> Ischlerbahnstraße 14, 5301 Eugendorf
> Tel: +43 662 879512 Fax: +43 662 875960
> IP-Tel: +43 780 commpany (26667269)
> Email: c.fuerstal...@commpany.at
> -BEGIN PGP SIGNATURE-
> Version: GnuPG v1.4.9 (MingW32)
>
> iEYEARECAAYFAknkvqoACgkQR0exH8dhr/ZqRACfV7KLoTMl9RgH0QNIPiJ/Gq9G
> 5dcAoIVK3L7pxTBLZrDi+kJGpOCPVa47
> =hEGE
> -END PGP SIGNATURE-
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
--
Atis Lezdins,
VoIP Project Manager / Developer,
IQ Labs Inc,
a...@iq-labs.net
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ring All Queue
At least in version 1.6.0.x you can specify a macro to be executed when the
agent answers the queued call. This is an argument to the queue application.
Queue(queuename[,options[,URL][,announceoverride][,timeout][,AGI][,macro][,g
osub][,rule])
The optional macro parameter will run a macro on the calling party's channel
once they are connected to a queue member.
Here is what my Macro does:
exten => s,1,UserEvent(DidQueue,ActionID:${CfMC_ActionID} & ${UNIQUEID}
& ${CHANNEL} & ${CfMC_AgentToUse} & ${CfMC_DialInfo} &
${CfMC_QueueToUse} & ${MEMBERINTERFACE} & ${MEMBERNAME})
${MEMBERINTERFACE} and ${MEMBERNAME} have info about the agent that answered
the call.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
> From: Luis Morales
> Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
>
> Date: Tue, 14 Apr 2009 13:17:49 -0430
> To: Asterisk Users Mailing List - Non-Commercial Discussion
>
> Subject: Re: [asterisk-users] Ring All Queue
>
> Try to parser queue_log file in real time and catch the event CONNECT
>
> Regards,
>
> Luis Morales
>
> On Tue, Apr 14, 2009 at 12:44 PM, Ryan M. Colbert
> wrote:
>> Is there a way in the dialplan to figure out which agent in a ring all queue
>> answered a line? I¹d like to take specific action based on the agent upon
>> hangup.
>>
>>
>>
>> Ryan M. Colbert
>> Director of Information Technology
>> Rissman, Barrett, Hurt,
>>
>> Donahue & McLain, P.A.
>> 201 E. Pine Street, Suite 1500
>> Orlando, FL 32801
>> (407) 517-3105 Direct Telephone
>> (407) 839-0120 - Main Office
>> (407) 841-9726 Fax
>> http://www.rissman.com/
>>
>>
>>
>> ___
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
>
> --
> --
> ---
> Luis Morales
> Consultor de Tecnologia
> Cel: +(58)416-4242091
> --
> ---
> "Empieza por hacer lo necesario, luego lo que es posible... y de
> pronto estarás haciendo lo imposible"
>
> Leonardo Da'Vinci
> --
> ---
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk-beginner : cannot make phonecalls using Asterisk
There is something wrong with my IPtables !!! When i do : service iptables stop I see my phones register on the CLI !! I can place a call and the phone rings !! I see a whole lot of SIP-requests on the CLI with SDP-message in body !! That's good news... What is wrong with my IPtables-rule I've added in /etc/sysconfig/iptables ??? [r...@asterisk sysconfig]# cat iptables # Firewall configuration written by system-config-securitylevel # Manual customization of this file is not recommended. *filter :INPUT ACCEPT [0:0] :FORWARD ACCEPT [0:0] :OUTPUT ACCEPT [0:0] :RH-Firewall-1-INPUT - [0:0] -A INPUT -j RH-Firewall-1-INPUT -A FORWARD -j RH-Firewall-1-INPUT -A RH-Firewall-1-INPUT -i lo -j ACCEPT -A RH-Firewall-1-INPUT -p icmp --icmp-type any -j ACCEPT -A RH-Firewall-1-INPUT -p 50 -j ACCEPT -A RH-Firewall-1-INPUT -p 51 -j ACCEPT -A RH-Firewall-1-INPUT -p udp --dport 5353 -d 224.0.0.251 -j ACCEPT -A RH-Firewall-1-INPUT -p udp -m udp --dport 631 -j ACCEPT -A RH-Firewall-1-INPUT -p tcp -m tcp --dport 631 -j ACCEPT -A RH-Firewall-1-INPUT -m state --state ESTABLISHED,RELATED -j ACCEPT -A RH-Firewall-1-INPUT -m state --state NEW -m tcp -p tcp --dport 22 -j ACCEPT -A RH-Firewall-1-INPUT -j REJECT --reject-with icmp-host-prohibited -A RH-Firewall-1-INPUT -p udp -m udp --dport 5060 -j ACCEPT COMMIT Greetingz, Jonas. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk-beginner : cannot make phonecalls using Asterisk
I will summarize everything again and try to answer all the questions asked while I was away. First I stop Asterisk : [r...@asterisk asterisk]# /usr/sbin/asterisk -r Asterisk 1.4.24, Copyright (C) 1999 - 2008 Digium, Inc. and others. Created by Mark Spencer Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. = Connected to Asterisk 1.4.24 currently running on asterisk (pid = 3189) Verbosity is at least 3 asterisk*CLI> stop now asterisk*CLI> Disconnected from Asterisk server [r...@asterisk asterisk]# ps aux | grep asterisk avahi 3320 0.0 0.0 2588 1344 ?Ss 18:49 0:00 avahi-daemon: running [asterisk.local] root 3563 0.0 0.0 3912 676 pts/0S+ 19:11 0:00 grep asterisk Then I edit the files sip.conf and extensions.conf SIP.CONF [r...@asterisk asterisk]# cat sip.conf [general] context=default port=5060 bindaddr=192.168.4.248 srvlookup=yes disallow=all allow=ulaw allow=gsm allow=g711 [BT201] type=friend context=intern host=dynamic username=BT201 secret=testpaswoord ;canreinvite=yes [GXP1200] type=friend context=intern host=dynamic username=GXP1200 secret=testpaswoord ;canreinvite=yes EXTENSIONS.CONF [r...@asterisk asterisk]# cat extensions.conf [globals] [default] [intern] exten => 210,1,Dial(SIP/BT201,30) exten => 211,1,Dial(SIP/GXP1200,30) exten => 251,1,Answer() exten => 251,n,Echo() exten => 251,n,Hangup() Then I configure my SIP-phone grandstream BT201 : 1) I press menu > dhcp [on] 2) I press menu > IP-address > 192.168.4.144 3) I go to the webinterface via the above IP-address My settings : > tab account account name : BT201 SIP server : 192.168.4.248 Outbound proxy : 192.168.4.248 SIP user ID : BT201 Authenticate ID : BT201 Authenticate Password : testpaswoord Name : BT201 Use DNS SRV : no User ID is phone number : no SIP registration : yes Unregister on reboot : no Register expiration : 60 local SIP port : 5060 SIP transport : UDP Use RFC3581 Symmetric Routing : no NAT Traversal (STUN) : no SUBSCRIBE for MWI : no Proxy-Require : (nothing) > Update > Reboot Then I configure my SIP-phone grandstream GX1200 : 1) I press menu > status 2) IP-address : 192.168.4.180 3) I go to the webinterface via the above IP-address My settings : > tab account account 1 active : yes account name : GX1200 SIP server : 192.168.4.248 Outbound proxy : 192.168.4.248 SIP user ID : GX1200 Authenticate ID : GX1200 Authenticate Password : testpaswoord Name : GX1200 Use DNS SRV : no User ID is phone number : no SIP registration : yes Unregister on reboot : no Register expiration : 60 local SIP port : 5060 SIP transport : UDP Use RFC3581 Symmetric Routing : no NAT Traversal (STUN) : no SUBSCRIBE for MWI : no Proxy-Require : (nothing) Then I unplug the power of the Grandstream IP-telephones. I restart Asterisk on my server : [r...@asterisk asterisk]# /sbin/service asterisk start Starting asterisk: [ OK ] [r...@asterisk asterisk]# /usr/sbin/asterisk -vvr Asterisk 1.4.24, Copyright (C) 1999 - 2008 Digium, Inc. and others. Created by Mark Spencer Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. = == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing '/etc/asterisk/extconfig.conf': Found Connected to Asterisk 1.4.24 currently running on asterisk (pid = 3683) Verbosity was 3 and is now 34 asterisk*CLI> I wait a while but no output on the CLI... Then I give some commands : asterisk*CLI> sip show peers Name/username HostDyn Nat ACL Port Status GXP1200/GXP1200(Unspecified)D 0 Unmonitored BT201/BT201(Unspecified)D 0 Unmonitored 2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 2 offline] asterisk*CLI> sip debug SIP Debugging enabled The 'sip debug' command is deprecated and will be removed in a future release. Please use 'sip set debug' instead. Then I power back on my Grandstream IP-telephones. Nothing happens on the CLI... asterisk*CLI> sip show peers Name/username HostDyn Nat ACL Port Status GXP1200/GXP1200(Unspecified)D 0 Unmonitored BT201/BT201(Unspecified)D 0 Unmonitored 2 sip peers [Monitored: 0 online, 0 offline Unmoni
Re: [asterisk-users] MOH
Khaled W. Chehab wrote: > Dear Ben, > > I tried a lot ,Kindly can you give me an example on how to do that using a > macro. > > Remove the 'm' out of your dial command: > > Doug > > I'm not Ben, but I'll answer. Shows us what your macro looks like and we'll chime in with some pointers. Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety." ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ring All Queue
Try to parser queue_log file in real time and catch the event CONNECT Regards, Luis Morales On Tue, Apr 14, 2009 at 12:44 PM, Ryan M. Colbert wrote: > Is there a way in the dialplan to figure out which agent in a ring all queue > answered a line? I’d like to take specific action based on the agent upon > hangup. > > > > Ryan M. Colbert > Director of Information Technology > Rissman, Barrett, Hurt, > > Donahue & McLain, P.A. > 201 E. Pine Street, Suite 1500 > Orlando, FL 32801 > (407) 517-3105 – Direct Telephone > (407) 839-0120 - Main Office > (407) 841-9726 – Fax > http://www.rissman.com/ > > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- - Luis Morales Consultor de Tecnologia Cel: +(58)416-4242091 - "Empieza por hacer lo necesario, luego lo que es posible... y de pronto estarás haciendo lo imposible" Leonardo Da'Vinci - ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ring All Queue
I have a same problem i add to the question Francisco Ryan M. Colbert wrote: > > Is there a way in the dialplan to figure out which agent in a ring all > queue answered a line? I’d like to take specific action based on the > agent upon hangup. > > Ryan M. Colbert > Director of Information Technology > Rissman, Barrett, Hurt, > > Donahue & McLain, P.A. > 201 E. Pine Street, Suite 1500 > Orlando, FL 32801 > (407) 517-3105 – Direct Telephone > (407) 839-0120 - Main Office > (407) 841-9726 – Fax > http://www.rissman.com/ > > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Ring All Queue
Is there a way in the dialplan to figure out which agent in a ring all queue answered a line? I'd like to take specific action based on the agent upon hangup. Ryan M. Colbert Director of Information Technology Rissman, Barrett, Hurt, Donahue & McLain, P.A. 201 E. Pine Street, Suite 1500 Orlando, FL 32801 (407) 517-3105 - Direct Telephone (407) 839-0120 - Main Office (407) 841-9726 - Fax http://www.rissman.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Exit Dial Application
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi,
Thanks for your replay. But this can only be done before or after the dial, but
I wanna do it during the dial, when user A is waiting for user B, answering the
phone. This should be possible, right?
I hope anyone knows if this is possible.
Chris...
Danny Nicholas schrieb:
> I'd change callback to this
> [callback]
> Exten => s,1,Playback(press5msg)
> Exten => s,n,Waitexten(5)
> Exten => s,n,Hangup
> exten => 5,1,agi(str_concat.sh)
> exten => 5,n,Hangup
>
> This will play a message, wait 5 seconds for user to press 5, then hangup if
> they don't.
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Christoph
> Fuerstaller
> Sent: Tuesday, April 14, 2009 5:04 AM
> To: Asterisk Users Mailing List
> Subject: [asterisk-users] Exit Dial Application
>
> Hi,
>
> I' try to implement an automatic callback mechanism, just for local SIP
> calls.. Callback
> on busy and on no answer. If the other party doen't answer, it should be
> possible to press
> 5 to place an callback.
>
> Here is my dial:
> exten => _X.,1,Set(EXITCONTEXT=callback)
> exten => _X.,n,Dial(${DIALNUM},${ARG2},dtT)
>
> And here the script for callback.
> [callback]
> exten => 5,1,agi(str_concat.sh)
> exten => 5,n,Hangup
>
> If I call someone and press 5, nothing happens. What could be a problem?
> DTMFmode is RFC2833 for all SIP Accounts. DTMF's are transmitted correctly,
> I can enter
> the voicmail menue.
>
> I'm using Asterisk 1.4.21.1.
>
> Any successions are very appreciated.
>
> Chris...
___
- -- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
- -- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
- --
commpany dialog solutions gmbh
Dipl.-Ing.(FH) Christoph Fürstaller
IP-Communications
Ischlerbahnstraße 14, 5301 Eugendorf
Tel: +43 662 879512 Fax: +43 662 875960
IP-Tel: +43 780 commpany (26667269)
Email: c.fuerstal...@commpany.at
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.9 (MingW32)
iEYEARECAAYFAknkvqoACgkQR0exH8dhr/ZqRACfV7KLoTMl9RgH0QNIPiJ/Gq9G
5dcAoIVK3L7pxTBLZrDi+kJGpOCPVa47
=hEGE
-END PGP SIGNATURE-
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] T38modem in loopback mode does not work on asterisk 1.4.20.1
David Backeberg wrote: > What I was specifically getting at in the context of that response was > a comparison of dynamic modem pool versus fixed-size modem pool. When > faced with the choice between a fixed-size modem pool or one that > would grow or shrink dynamically with demand, I think the dynamic is a > better choice. That is my opinion, based on my particular style of > usage, and I am certainly grateful for the existence of Hylafax. > [..] > The dynamic modem pool approach is nice in a number of ways, but has a key downside - there isn't an effect constraint on the upper bound of concurrent instances. You can keep starting them until you drag the machine to its knees. The same is true for other apps, of course, and any meaningful constraint would need to be based on the aggregate load they produce. On the upside, at least one fast modern machine have been seen steadily running 500 instances. :-) Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Vacation reply
Hi Friend, How are you doing recently? I would like to introduce you a very good company which I know. Their website is www.myewell.com. They can offer you all kinds of Electronic products like laptops, gps,TV LCD,cell phones,ps3,MP3/4, etcPlease take some time to have a check, There must have something you'd like to buy.Their contact email: myew...@vip.188.com Hope you have a good mood in shopping from their company! Best Regards aymen ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Vacation reply
Hi Friend, How are you doing recently? I would like to introduce you a very good company which I know. Their website is www.myewell.com. They can offer you all kinds of Electronic products like laptops, gps,TV LCD,cell phones,ps3,MP3/4, etcPlease take some time to have a check, There must have something you'd like to buy.Their contact email: myew...@vip.188.com Hope you have a good mood in shopping from their company! Best Regards aymen ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MOH
Dear Ben, I tried a lot ,Kindly can you give me an example on how to do that using a macro. Regards -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle Sent: Tuesday, April 14, 2009 5:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] MOH Khaled W. Chehab wrote: > Dears > > -How can I stop MOH when status of the dial is ringing and let the user hear > the Ring Back Tone from the termination Gateway. > Remove the 'm' out of your dial command: m([class]) - Provide hold music to the calling party until a requested channel answers. A specific MusicOnHold class can be specified. Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety." ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] OT - snom phone question
Sorry - this is a bit off topic, but there is almost certainly someone here who will know the answer... Perhaps even a snom employee :) In recent snom firmware releases, the following sequence always causes a call to be sent from line 'n' Receive call on Line 'n' (where n > 1) Press Hold Dial new call The new call originates on Line 'n' In older firmware versions, the 2nd call would always originate from line 1. Is the default line selectable from the phone's configuration anywhere? I completly understand the logic behind the new behaviour, but some users prefer the old way, and I cannot find a setting for it. Basically, in an office environment, using line 1 is more meaningful. In a Centrex environment, using line 'n' is often the desired choice. Thanks, Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Vacation reply
Hi Friend, How are you doing recently? I would like to introduce you a very good company which I know. Their website is www.myewell.com. They can offer you all kinds of Electronic products like laptops, gps,TV LCD,cell phones,ps3,MP3/4, etcPlease take some time to have a check, There must have something you'd like to buy.Their contact email: myew...@vip.188.com Hope you have a good mood in shopping from their company! Best Regards aymen ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Vacation reply
Hi Friend, How are you doing recently? I would like to introduce you a very good company which I know. Their website is www.myewell.com. They can offer you all kinds of Electronic products like laptops, gps,TV LCD,cell phones,ps3,MP3/4, etcPlease take some time to have a check, There must have something you'd like to buy.Their contact email: myew...@vip.188.com Hope you have a good mood in shopping from their company! Best Regards aymen ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk-beginner : cannot make phonecalls usingAsterisk
Let's just simplify this a LOT: Your phones have no dialtone. This means they are not registering with asterisk. I see in your sip.conf, for both you phones, you have: host=X.X.X.X If you specify an address here, your phones will not register. Instead, to make your phones register, set it to: host=dynamic (It does not matter if the phones are configured dynamically via DHCP or statically configured, but they do need to be configured to try and register to asterisk) For both phones, you might also want to add: qualify=yes This will monitor whether or not the phones are in contact with asterisk. You can view this with "sip show peers". Once you've gotten the phones to register with asterisk, THEN try having them call one another. - Noah ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Vacation reply
Hi Friend, How are you doing recently? I would like to introduce you a very good company which I know. Their website is www.myewell.com. They can offer you all kinds of Electronic products like laptops, gps,TV LCD,cell phones,ps3,MP3/4, etcPlease take some time to have a check, There must have something you'd like to buy.Their contact email: myew...@vip.188.com Hope you have a good mood in shopping from their company! Best Regards aymen ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] T38modem in loopback mode does not work on asterisk 1.4.20.1
Steve Underwood wrote: > Lee Howard wrote: > >> David Backeberg wrote: >> >> >>> It may be possible to use hylafax, but >>> I don't know how or why you would. >>> >>> >> The reason *why* is generally due to support issues. >> >> For one, HylaFAX probably has a better T.30 implementation in its Class >> 1 driver than does app_fax. At least that historically has been true. >> I welcome the day when all fax applications perform as well as HylaFAX >> has since its 4.2.x days. >> >> > It used to be true, but I suspect there is little in it when you use > spandsp-0.0.6pre7 or later. :-) That's wonderful news. :-) Thanks, Lee. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] duration of rfc2833 generated dtmf
on Tuesday 04/14/2009 Kristian Kielhofner(kristian.kielhof...@gmail.com) wrote > On Mon, Apr 13, 2009 at 5:32 PM, John covici wrote: > > Hi. I have a SIP provider which wants RFC2833 for the dtmfmode, > > however I would like to increase the duration of the tone, its pretty > > short and some IVR's are unhappy or don't detect it. I did poke > > around, but it looks like when RFC2833 is used, it actually generates > > rtp packets of some sort, so I have no idea how to increase that > > duration. > > > > Any assistance would be appreciated. > > > > -- > > Your life is like a penny. You're going to lose it. The question is: > > How do > > you spend it? > > > > John Covici > > cov...@ccs.covici.com > > John, > > Assuming this is Asterisk 1.4 or later... The duration used by > Asterisk is the same duration sent from the phone. The duration of > those DTMF key presses should match the time the user is holding down > the key. What type(s) of phones are these? You should also look into > using Asterisk 1.4.24.1 or later (if you aren't already). There have > been many improvements to the RTP code to better handle quirks with > the equipment (especially Sonus) used by various providers. > > Assuming your provider is to spec (and so is your phone) your > provider should not be complaining that the duration of your DTMF key > presses are too short... > > With that being said AFAIK there is no way to specify a minimum > duration for an RFC 2833 DTMF in Asterisk on a bridged channel. OK, thanks for that info -- but it seems to me no matter how long I press the keys on the phone, (connected to a Digium board) the other end gets the same duration. Now, the problems I run into are not dialing the phone number, but dtmf on the call such as an IVR. Some of them don't like what seems to be a too short key press, whereas if I call the same number from the cell phone, there is no problem. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Vacation reply
Hi Friend, How are you doing recently? I would like to introduce you a very good company which I know. Their website is www.myewell.com. They can offer you all kinds of Electronic products like laptops, gps,TV LCD,cell phones,ps3,MP3/4, etcPlease take some time to have a check, There must have something you'd like to buy.Their contact email: myew...@vip.188.com Hope you have a good mood in shopping from their company! Best Regards aymen ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Vacation reply
Hi Friend, How are you doing recently? I would like to introduce you a very good company which I know. Their website is www.myewell.com. They can offer you all kinds of Electronic products like laptops, gps,TV LCD,cell phones,ps3,MP3/4, etcPlease take some time to have a check, There must have something you'd like to buy.Their contact email: myew...@vip.188.com Hope you have a good mood in shopping from their company! Best Regards aymen ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] duration of rfc2833 generated dtmf
On Mon, Apr 13, 2009 at 5:32 PM, John covici wrote: > Hi. I have a SIP provider which wants RFC2833 for the dtmfmode, > however I would like to increase the duration of the tone, its pretty > short and some IVR's are unhappy or don't detect it. I did poke > around, but it looks like when RFC2833 is used, it actually generates > rtp packets of some sort, so I have no idea how to increase that > duration. > > Any assistance would be appreciated. > > -- > Your life is like a penny. You're going to lose it. The question is: > How do > you spend it? > > John Covici > cov...@ccs.covici.com John, Assuming this is Asterisk 1.4 or later... The duration used by Asterisk is the same duration sent from the phone. The duration of those DTMF key presses should match the time the user is holding down the key. What type(s) of phones are these? You should also look into using Asterisk 1.4.24.1 or later (if you aren't already). There have been many improvements to the RTP code to better handle quirks with the equipment (especially Sonus) used by various providers. Assuming your provider is to spec (and so is your phone) your provider should not be complaining that the duration of your DTMF key presses are too short... With that being said AFAIK there is no way to specify a minimum duration for an RFC 2833 DTMF in Asterisk on a bridged channel. -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Vacation reply
awerf...@hotmail.com wrote: > > Hi Friend, > Oh boy, this is fun. Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety." ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk-beginner : cannot make phonecalls usingAsterisk
Since we don't know if he's having his phone's register, why not have asterisk try to register them? If not, just another dumb suggestion from me... -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards Sent: Tuesday, April 14, 2009 11:06 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk-beginner : cannot make phonecalls usingAsterisk On Tue, 14 Apr 2009, Danny Nicholas wrote: > Put register=yes in the BT201 and GXP1200 contexts of sip.conf I admit to being a 1.2 Luddite, but isn't "register" asking your Asterisk server to register with another endpoint? As in: register => ::[authid]@/ Or is this some sort of 1.4/1.6 overloading of an existing keyword? Thanks in advance, Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What means? Correct auth, but based on stale nonce received
http://lists.digium.com/pipermail/asterisk-users/2005-July/110220.html It means that a SIP device is re-using an old authentication challenge. If it still registers and can place calls, there's no problem to worry about. It's just a warning. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tiago Durante Sent: Tuesday, April 14, 2009 11:06 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] What means? Correct auth,but based on stale nonce received Hi masters! I've this Asterisk 1.4.15 running. yesterday I had to change the firewall schema that I had before. I use to have a FW that would be my network FW/Proxy and do the NATs for Asterisk. This FW was receiving too many requests from my LAN and it was making the Asterisk 'cut' the calls or reach very high latency. Yesterday I added a new FW just for this Asterisk. The same configuration as the old firewall, loading the same modules, same NATs. But now some ATAs (sip) can't register against this Asterisk and the ones that can generates these messages: [Apr 14 13:01:45] NOTICE[29235]: chan_sip.c:8375 check_auth: Correct auth, but based on stale nonce received from 'sip:xx...@200.x.x.x:5060' [Apr 14 13:01:45] NOTICE[29235]: chan_sip.c:8375 check_auth: Correct auth, but based on stale nonce received from 'sip:xx...@200.x.x.x:5060' Does any one know why this happens? Thank you!!! -- Tiago Durante ,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,., Perseverance is the hard work you do after you get tired of doing the hard work you already did. -- Newt Gingrich ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Vacation reply
Hi Friend, How are you doing recently? I would like to introduce you a very good company which I know. Their website is www.myewell.com. They can offer you all kinds of Electronic products like laptops, gps,TV LCD,cell phones,ps3,MP3/4, etcPlease take some time to have a check, There must have something you'd like to buy.Their contact email: myew...@vip.188.com Hope you have a good mood in shopping from their company! Best Regards aymen ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Vacation reply
Hi Friend, How are you doing recently? I would like to introduce you a very good company which I know. Their website is www.myewell.com. They can offer you all kinds of Electronic products like laptops, gps,TV LCD,cell phones,ps3,MP3/4, etcPlease take some time to have a check, There must have something you'd like to buy.Their contact email: myew...@vip.188.com Hope you have a good mood in shopping from their company! Best Regards aymen ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Vacation reply
Hi Friend, How are you doing recently? I would like to introduce you a very good company which I know. Their website is www.myewell.com. They can offer you all kinds of Electronic products like laptops, gps,TV LCD,cell phones,ps3,MP3/4, etcPlease take some time to have a check, There must have something you'd like to buy.Their contact email: myew...@vip.188.com Hope you have a good mood in shopping from their company! Best Regards aymen ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] What means? Correct auth, but based on stale nonce received
Hi masters! I've this Asterisk 1.4.15 running. yesterday I had to change the firewall schema that I had before. I use to have a FW that would be my network FW/Proxy and do the NATs for Asterisk. This FW was receiving too many requests from my LAN and it was making the Asterisk 'cut' the calls or reach very high latency. Yesterday I added a new FW just for this Asterisk. The same configuration as the old firewall, loading the same modules, same NATs. But now some ATAs (sip) can't register against this Asterisk and the ones that can generates these messages: [Apr 14 13:01:45] NOTICE[29235]: chan_sip.c:8375 check_auth: Correct auth, but based on stale nonce received from 'sip:xx...@200.x.x.x:5060' [Apr 14 13:01:45] NOTICE[29235]: chan_sip.c:8375 check_auth: Correct auth, but based on stale nonce received from 'sip:xx...@200.x.x.x:5060' Does any one know why this happens? Thank you!!! -- Tiago Durante ,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,., Perseverance is the hard work you do after you get tired of doing the hard work you already did. -- Newt Gingrich ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk-beginner : cannot make phonecalls usingAsterisk
On Tue, 14 Apr 2009, Danny Nicholas wrote: > Put register=yes in the BT201 and GXP1200 contexts of sip.conf I admit to being a 1.2 Luddite, but isn't "register" asking your Asterisk server to register with another endpoint? As in: register => ::[authid]@/ Or is this some sort of 1.4/1.6 overloading of an existing keyword? Thanks in advance, Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Vacation reply
Hi Friend, How are you doing recently? I would like to introduce you a very good company which I know. Their website is www.myewell.com. They can offer you all kinds of Electronic products like laptops, gps,TV LCD,cell phones,ps3,MP3/4, etcPlease take some time to have a check, There must have something you'd like to buy.Their contact email: myew...@vip.188.com Hope you have a good mood in shopping from their company! Best Regards aymen ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] .GSM -> .WAV (or ,MP3) Conversion
Tim Dobson wrote: > 1239101491.30.gsm -ql -r 64000 -t wav 1239101491.30.conv.wav resample > as ffmpeg borks at it: Gah I meant sox 1239101491.30.gsm -ql -r 64000 -t wav 1239101491.30.conv.wav resample -- Thanks everyone! Really appreciate it - all the documentation via google is to convert *to* gsm... nothing says anything about *from* it! Cheers! Tim -- www.tdobson.net If each of us have one object, and we exchange them, then each of us still has one object. If each of us have one idea, and we exchange them, then each of us now has two ideas. - George Bernard Shaw ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] T38modem in loopback mode does not work on asterisk 1.4.20.1
Lee Howard wrote: > David Backeberg wrote: > >> It may be possible to use hylafax, but >> I don't know how or why you would. >> > > The reason *why* is generally due to support issues. > > For one, HylaFAX probably has a better T.30 implementation in its Class > 1 driver than does app_fax. At least that historically has been true. > I welcome the day when all fax applications perform as well as HylaFAX > has since its 4.2.x days. > It used to be true, but I suspect there is little in it when you use spandsp-0.0.6pre7 or later. :-) Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Vacation reply
Hi Friend, How are you doing recently? I would like to introduce you a very good company which I know. Their website is www.myewell.com. They can offer you all kinds of Electronic products like laptops, gps,TV LCD,cell phones,ps3,MP3/4, etcPlease take some time to have a check, There must have something you'd like to buy.Their contact email: myew...@vip.188.com Hope you have a good mood in shopping from their company! Best Regards aymen ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dynamic menus in dialplan
Steve Edwards had (IMO) the best answer to this. Here is an example of
how-to in a regular dialplan:
[sat]
exten => s,1(start),Noop(in test Section)
exten => s,n,AGI(satver.agi|)
exten => s,n,Set(Play1=record/silence)
exten => s,n,Set(Play2=record/silence)
exten => s,n,Set(Press1=record/silence)
exten => s,n,Set(Press2=record/silence)
exten => s,n,Set(Play3=record/silence)
exten => s,n,Set(Press3=record/silence)
exten => s,n,Set(PRESS=0)
exten => s,n(s1check),Gotoif($["${sat1}" != "TRUE"]?s2check)
exten => s,n,Set(Play1=record/nsfmenu)
exten => s,n,Set(Press1=record/press1)
exten => s,n,Set(PRESS=1)
exten => s,n,Goto(sat|s|s2check)
exten => s,n(s2check),Gotoif($["${sat2}" != "TRUE"]?s3check)
.
exten =>
s,n(allchecked),Background(${Play1}&${Press1}&${Play2}&${Press2}&${Play3}&${
Press3}&${Play4}&${Press4})
exten => s,n,WaitExten(10,m)
exten => s,n,Set(TRY=$[${TRY} + 1])
exten => s,n,Gotoif($["${TRY}" = "3"]?expired)
exten => s,n,Goto(sat|s|start)
exten => s,n(expired),playback(vm-goodbye)
exten => s,n,Hangup
exten => i,1,Set(TRY=$[${TRY} + 1])
exten => i,n,Verbose(Invalid Try # ${TRY} )
exten => i,n,Gotoif($["${TRY}" = "3"]?expired)
exten => i,n,Background(invalid)
exten => i,n,Goto(sat|s|start)
exten => t,1,Goto(sat|s|start)
exten => 1,1,Gotoif($["${sat1}" = "TRUE"]?sat1|s|1)
exten => 1,n,Gotoif($["${sat2}" = "TRUE"]?sat1|s|1)
exten => 2,1,Gotoif($["${sat2}" = "TRUE"]?sat2|s|1)
exten => 2,n,Gotoif($["${sat3}" = "TRUE"]?sat3|s|1)
[sat1]
Sat1 stuff
[sat2]
Sat2 stuff
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David fire
Sent: Monday, April 13, 2009 9:54 PM
To: eric.f...@gmail.com; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [asterisk-users] dynamic menus in dialplan
hi
you can use AGI or a database internal or external
then if you know all the satellites and are a few you can
if(${SAT1}=1)
playback(SAT1)
if(${SAT2}=1)
playback(SAT2)
.
.
.
or you can use an agi
David
2009/4/14 Eric Fort
I have an application that needs to vary the menu choices available based
upon the availability of an external resource at a given time. What I have
in mind is a system that can uplink a user to one of many different
satellites. Due to the nature of orbital mechanics a satellite may be out
of range at any given time. I only want to present a menu of available
satellites. I can query an external program for a list of available
satellites, but how can I use that list to present menu options for
selection? What's the best way of doing this? Does anyone know of similar
examples?
Thanks,
Eric
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
(\__/)
(='.'=)This is Bunny. Copy and paste bunny into your
(")_(")signature to help him gain world domination.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Vacation reply
Hi Friend, How are you doing recently? I would like to introduce you a very good company which I know. Their website is www.myewell.com. They can offer you all kinds of Electronic products like laptops, gps,TV LCD,cell phones,ps3,MP3/4, etcPlease take some time to have a check, There must have something you'd like to buy.Their contact email: myew...@vip.188.com Hope you have a good mood in shopping from their company! Best Regards aymen ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Vacation reply
Hi Friend, How are you doing recently? I would like to introduce you a very good company which I know. Their website is www.myewell.com. They can offer you all kinds of Electronic products like laptops, gps,TV LCD,cell phones,ps3,MP3/4, etcPlease take some time to have a check, There must have something you'd like to buy.Their contact email: myew...@vip.188.com Hope you have a good mood in shopping from their company! Best Regards aymen ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Vacation reply
Hi Friend, How are you doing recently? I would like to introduce you a very good company which I know. Their website is www.myewell.com. They can offer you all kinds of Electronic products like laptops, gps,TV LCD,cell phones,ps3,MP3/4, etcPlease take some time to have a check, There must have something you'd like to buy.Their contact email: myew...@vip.188.com Hope you have a good mood in shopping from their company! Best Regards aymen ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk-beginner : cannot make phonecalls using Asterisk
just for a test, run "service iptables stop" as root on the asterisk server and then reboot your phones. after that, try again and see if the phones are making communications with asterisk. you can turn the firewall back on with "service iptables start" jonas kellens wrote: > Hi there, > > this is the first time that I'm building an Asterisk-server. > > I have compiled Asterisk together with Zaptel on an CentOS 5.3-system. > Zaptel is for later, when configuring the POTS-line. Now first internal > communication with SIP. > > Thought it would go easier... > > I have 2 Grandstream IP-phones : BT-201 and GXP-1200. > > These are my settings : > > sip.conf : > /[r...@asterisk asterisk]# cat sip.conf/ > /[general]/ > /bindport=5060/ > /bindaddr = 0.0.0.0/ > > /[BT201]/ > /type=friend/ > /context=intern/ > /host=192.168.4.210/ > /secret=testpaswoord/ > > /[GXP1200]/ > /type=friend/ > /context=intern/ > /host=192.168.4.211/ > /secret=testpaswoord/ > extensions.conf : > /[r...@asterisk asterisk]# cat extensions.conf/ > /[intern]/ > /exten => 210,1,Dial(SIP/BT201)/ > /exten => 211,1,Dial(SIP/GXP1200)/ > Asterisk CLI shows me : > /asterisk*CLI> sip reload/ > /Reloading SIP/ > / == Parsing '/etc/asterisk/sip.conf': Found/ > / == Parsing '/etc/asterisk/users.conf': Found/ > / == Parsing '/etc/asterisk/sip_notify.conf': Found/ > /asterisk*CLI> sip show peers/ > /Name/username HostDyn Nat ACL Port Status > / > /GXP1200192.168.4.211 5060 > Unmonitored / > /BT201 192.168.4.210 5060 > Unmonitored / > /2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 2 online, 0 > offline]/ > > /asterisk*CLI> dialplan show intern/ > /[ Context 'intern' created by 'pbx_config' ]/ > / '210' => 1. Dial(SIP/BT201) > [pbx_config]/ > / '211' => 1. Dial(SIP/GXP1200) > [pbx_config]/ > > I pick up the phone of the BT201 and dial 211... nothing happens. > I pick up the phone of the GXP1200 and dial 210... nothing happens. > > I would love to have your feedback on this. Where could this problem be > situated ? > > I notice (on the Asterisk CLI) that my SIP-phones do not register. They > have a fixed IP and there account information is set via the web interface. > > Greetingz, > Jonas. > > > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] T38modem in loopback mode does not work on asterisk 1.4.20.1
On Tue, Apr 14, 2009 at 10:46 AM, Lee Howard wrote: > David Backeberg wrote: >> It may be possible to use hylafax, but >> I don't know how or why you would. > > The reason *why* is generally due to support issues. What I was specifically getting at in the context of that response was a comparison of dynamic modem pool versus fixed-size modem pool. When faced with the choice between a fixed-size modem pool or one that would grow or shrink dynamically with demand, I think the dynamic is a better choice. That is my opinion, based on my particular style of usage, and I am certainly grateful for the existence of Hylafax. The other thing I was getting at was complexity of a multi-layer software stack in loopback mode when it comes to trying to figure out where the problem could be. One of my suggestions was to separate the sender and receiver to separate machines, and my other suggestion was to reduce the layers of software involved. As you know from having written Hylafax, it's designed for hardware modems and does things like letting you take your fax device online/offline, as well as taking it down for five seconds post-fax to do a reset on the line. These things are all documented, they all have debugging tools, and ways to view modem state. But for a beginning person trying to set up a fax for the first time I feel like it adds complexity to what can already be a complex task. Thus I suggested trying without. As I've been debugging faxes it seems more and more like fax failures are due to poorly-thought-through enterprise voip implementations, or intermittent line noise, or poor implementations of fax protocols in proprietary fax machine hardware. I think the Hylafax versus app_fax thing is really six of one versus a half-dozen of the other, but both are useful data points in debugging a failing fax configuration. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Exit Dial Application
I'd change callback to this
[callback]
Exten => s,1,Playback(press5msg)
Exten => s,n,Waitexten(5)
Exten => s,n,Hangup
exten => 5,1,agi(str_concat.sh)
exten => 5,n,Hangup
This will play a message, wait 5 seconds for user to press 5, then hangup if
they don't.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Christoph
Fuerstaller
Sent: Tuesday, April 14, 2009 5:04 AM
To: Asterisk Users Mailing List
Subject: [asterisk-users] Exit Dial Application
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi,
I' try to implement an automatic callback mechanism, just for local SIP
calls.. Callback
on busy and on no answer. If the other party doen't answer, it should be
possible to press
5 to place an callback.
Here is my dial:
exten => _X.,1,Set(EXITCONTEXT=callback)
exten => _X.,n,Dial(${DIALNUM},${ARG2},dtT)
And here the script for callback.
[callback]
exten => 5,1,agi(str_concat.sh)
exten => 5,n,Hangup
If I call someone and press 5, nothing happens. What could be a problem?
DTMFmode is RFC2833 for all SIP Accounts. DTMF's are transmitted correctly,
I can enter
the voicmail menue.
I'm using Asterisk 1.4.21.1.
Any successions are very appreciated.
Chris...
- --
commpany dialog solutions gmbh
Dipl.-Ing.(FH) Christoph Fürstaller
IP-Communications
Ischlerbahnstraße 14, 5301 Eugendorf
Tel: +43 662 879512 Fax: +43 662 875960
IP-Tel: +43 780 commpany (26667269)
Email: c.fuerstal...@commpany.at
-BEGIN PGP SIGNATURE-
Version: GnuPG v2.0.11 (GNU/Linux)
iEYEARECAAYFAknkX5UACgkQR0exH8dhr/bIpgCffDCaHgDO6bWltTQHOajL63ZI
YTMAn0jDBdNOxsd5jjxBZ1yJ2J9HcCR5
=K4sI
-END PGP SIGNATURE-
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] duration of rfc2833 generated dtmf
To the best of my knowledge, the only way for you to control the duration sent to the PSTN lines is for you to be directly connected to the lines so you can set the tone duration in zapata.conf / dahdi.conf or to use inband signalling. One thing you might try is researching the "SipDtmfMode" command. It allows you to change the DTMF mode on an active channel. A suggestion might be to set up the dial command with the M() option that point to a Macro that changes the DTMF to INBAND once you are connected to the problem number. At least in theory, if your provider is expecting RFC2833 and they get inband, they should just ignore the inband signaling and pass it on as part of the audio stream. The only problem is that this may only work if you use uLaw or aLaw for your codec and I don't know exactly how to set the tone duration without having a zapata.conf or dahdi.conf entry. Even with one of those files, I don't know how Asterisk chooses to do the rfc2833 to inband translation or where it pulls the toneduration setting from if no PSTN interface is involved in the call. -Brent John covici wrote: > OK, thanks. If I could convince them to use info, would that be > better as far as the duration is concerned? > > > on Monday 04/13/2009 Brent Davidson(br...@texascountrytitle.com) wrote > > John covici wrote: > > > Hi. I have a SIP provider which wants RFC2833 for the dtmfmode, > > > however I would like to increase the duration of the tone, its pretty > > > short and some IVR's are unhappy or don't detect it. I did poke > > > around, but it looks like when RFC2833 is used, it actually generates > > > rtp packets of some sort, so I have no idea how to increase that > > > duration. > > > > > > Any assistance would be appreciated. > > > > > > > > > > If your provider insists on rfc2833, then their servers will be > > responsible for setting the tone duration sent to PSTN lines. > > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk-beginner : cannot make phonecalls usingAsterisk
Put register=yes in the BT201 and GXP1200 contexts of sip.conf _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of jonas kellens Sent: Monday, April 13, 2009 11:19 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk-beginner : cannot make phonecalls usingAsterisk Hi there, this is the first time that I'm building an Asterisk-server. I have compiled Asterisk together with Zaptel on an CentOS 5.3-system. Zaptel is for later, when configuring the POTS-line. Now first internal communication with SIP. Thought it would go easier... I have 2 Grandstream IP-phones : BT-201 and GXP-1200. These are my settings : sip.conf : [r...@asterisk asterisk]# cat sip.conf [general] bindport=5060 bindaddr = 0.0.0.0 [BT201] type=friend context=intern host=192.168.4.210 secret=testpaswoord [GXP1200] type=friend context=intern host=192.168.4.211 secret=testpaswoord extensions.conf : [r...@asterisk asterisk]# cat extensions.conf [intern] exten => 210,1,Dial(SIP/BT201) exten => 211,1,Dial(SIP/GXP1200) Asterisk CLI shows me : asterisk*CLI> sip reload Reloading SIP == Parsing '/etc/asterisk/sip.conf': Found == Parsing '/etc/asterisk/users.conf': Found == Parsing '/etc/asterisk/sip_notify.conf': Found asterisk*CLI> sip show peers Name/username HostDyn Nat ACL Port Status GXP1200192.168.4.211 5060 Unmonitored BT201 192.168.4.210 5060 Unmonitored 2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 2 online, 0 offline] asterisk*CLI> dialplan show intern [ Context 'intern' created by 'pbx_config' ] '210' => 1. Dial(SIP/BT201) [pbx_config] '211' => 1. Dial(SIP/GXP1200) [pbx_config] I pick up the phone of the BT201 and dial 211... nothing happens. I pick up the phone of the GXP1200 and dial 210... nothing happens. I would love to have your feedback on this. Where could this problem be situated ? I notice (on the Asterisk CLI) that my SIP-phones do not register. They have a fixed IP and there account information is set via the web interface. Greetingz, Jonas. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk-beginner : cannot make phonecalls using Asterisk
This is what I derive from your emails so far: 1. Asterisk is up and running 2. you have two SIP Peers that are running but not registered (sip show peers shows not monitored, phones don't go off-hook, dialtone) 3. I've experienced similar problem with a X-Lite softphone. When the phone properly registers with Asterisk, you will be good to go. Have you configured the phones to register with your Asterisk? (ID, password, IP, port?) _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of jonas kellens Sent: Monday, April 13, 2009 1:47 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk-beginner : cannot make phonecalls using Asterisk Barry, there is a 'send' button but pushing it before or after dialing '211' does not really change anything... I get no dial tone, no ring tone on the other phone and no output on the Asterisk CLI... I thought this would go easier... Don't know what is going on here. I followed the book "Asterisk, the future of telephony"... Thanks for your reply ! Greetingz, Jonas. On Mon, 2009-04-13 at 14:04 -0400, Barry L. Kline wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 jonas kellens wrote: > I pick up the phone, and dial 211 on the BT201. This is the Asterisk CLI : > > /Connected to Asterisk 1.4.24 currently running on asterisk (pid = 3895)/ > /Verbosity is at least 5/ > /asterisk*CLI> / > > Nothing is displayed... it stays that way... > > Jonas. Is there a "Send" button on that phone? It sounds to me as though the phone is still waiting for more digits. Barry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk-beginner : cannot make phonecalls using Asterisk
You are doing 210 (enter), 210(dial) or 210 (#)? You have to engage the dialer in some fashion. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of jonas kellens Sent: Monday, April 13, 2009 12:20 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk-beginner : cannot make phonecalls using Asterisk James, when I run Asterisk -vr and I enter 210 on one phone to call the other, nothing is displayed on the CommandLine... I know this is not right, just don't know what is wrong. I really need someone to guide me a bit... [r...@asterisk asterisk]# /usr/sbin/asterisk -vr Asterisk 1.4.24, Copyright (C) 1999 - 2008 Digium, Inc. and others. Created by Mark Spencer Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. = == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing '/etc/asterisk/extconfig.conf': Found Connected to Asterisk 1.4.24 currently running on asterisk (pid = 3895) Verbosity is at least 5 asterisk*CLI> sip reload Reloading SIP == Parsing '/etc/asterisk/sip.conf': Found == Parsing '/etc/asterisk/users.conf': Found == Parsing '/etc/asterisk/sip_notify.conf': Found asterisk*CLI> sip show peers Name/username HostDyn Nat ACL Port Status GXP1200/GXP1200192.168.4.211 5060 Unmonitored BT201/BT201192.168.4.210 5060 Unmonitored 2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 2 online, 0 offline] Thanks for your reply ! Jonas. On Mon, 2009-04-13 at 11:59 -0500, James A. Shigley wrote: What do you see when you run asterisk -r and dial 210 or 211 from one of the phones James Shigley ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] .GSM -> .WAV (or ,MP3) Conversion
In sox 14.0.1 the -1 is a one byte sample, -2 is a two byte sample, therefore the command is sampling one-in, two-out. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards Sent: Tuesday, April 14, 2009 9:08 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] .GSM -> .WAV (or ,MP3) Conversion On Tue, 14 Apr 2009, Arjan Kroon | Mobillion wrote: > I record the message in ULAW > > After that I call sox with this command: > /usr/bin/sox -c 1 -1 -t ul -r 8000 $in_fl -t wav -2 -r 8000 -c 1 > $wav_fl What are "-1" and "-2" for? Both sox 12.17.5 and 12.18.1 say they are invalid. Thanks in advance, Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk-beginner : cannot make phonecallsusing Asterisk
You are closing in. what does users.conf look like? _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of jonas kellens Sent: Monday, April 13, 2009 3:40 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk-beginner : cannot make phonecallsusing Asterisk Hi Tzafrir, yet with the first test, things get wrong : asterisk*CLI> logger show channels Channel Type StatusConfiguration --- --- /var/log/asterisk/messages File Enabled- Warning Notice Error Console Enabled- Warning Notice Error asterisk*CLI> asterisk*CLI> originate SIP/210 application playback demo-instruct [Apr 13 22:33:23] NOTICE[3481]: channel.c:3033 __ast_request_and_dial: Unable to request channel SIP/210 asterisk*CLI> Instead of naming the phone BT201, I've named it after its internal telephone number. For clearity for myself :-). But when I dial the IP-phone from the CLI, I get the output of above... Thank for your reply ! Jonas. On Mon, 2009-04-13 at 23:11 +0300, Tzafrir Cohen wrote: Hi On Mon, Apr 13, 2009 at 06:18:58PM +0200, jonas kellens wrote: > I pick up the phone of the BT201 and dial 211... nothing happens. > I pick up the phone of the GXP1200 and dial 210... nothing happens. > > I would love to have your feedback on this. Where could this problem be > situated ? Your basic mistake at troubleshooting this is trying to test two things at the same time. Let's test them separately. 1. A call from Asterisk to the phones: In the Asterisk CLI: originate SIP/BT201 application playback demo-instruct And the other one: originate SIP/GXP1200 application playback demo-instruct Alternatively, use the echo-test aplication: originate SIP/BT201 application echo 2. Next, test calling from the phones to Asterisk. Add those two extensions to [intern] exten => 250,1,Answer exten => 250,n,Playback(demo-instruct) exten => 250,n,Hangup exten => 251,1,Answer exten => 251,1,Echo exten => 251,1,Hangup Make sure you reload for that to take effect, and then try dialing 250 or 251. Another useful tools: 'sip debug'. It tends to generate a very noisy output that is normally not readable for mere mortals. However it does indicate that "something is happening". If you call from a remote SIP phone and there's nothing on the SIP debug, the problem is probably with the settings of the phone, as it is not getting to you. Last and not least: a sanity check as you "see nothing": what is the output of: 'logger show channels' ? <>___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MOH
Khaled W. Chehab wrote: > Dears > > -How can I stop MOH when status of the dial is ringing and let the user hear > the Ring Back Tone from the termination Gateway. > Remove the 'm' out of your dial command: m([class]) - Provide hold music to the calling party until a requested channel answers. A specific MusicOnHold class can be specified. Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety." ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] T38modem in loopback mode does not work on asterisk 1.4.20.1
David Backeberg wrote: > It may be possible to use hylafax, but > I don't know how or why you would. The reason *why* is generally due to support issues. For one, HylaFAX probably has a better T.30 implementation in its Class 1 driver than does app_fax. At least that historically has been true. I welcome the day when all fax applications perform as well as HylaFAX has since its 4.2.x days. Thanks, Lee. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ignoring time spent waiting in queue in CDR
Scott Gifford escribió: > Hello, > > I'm working on an Asterisk configuration for a call center, and they > bill based on the time spent talking to an agent, but not for any time > spent waiting in a queue. The CDR information contains the entire > duration of the call as billable seconds, including time spent waiting > in the queue. I would like the billable seconds to only include the > time spent actually talking to an agent. > > I am using Asterisk 1.4.18. > > The only way I have found so far is to correlate the CDRs with the > "CONNECT" queue records, figure out the end time of the call by adding > the CDR start time to the duration, then figure out the actual > duration by subtracting the time of the queue "CONNECT" record. That > seems messy and error-prone, and I'm hoping there's a better way. > Why don't you just make your billing statistics from the queue log? Assuming that you upload the queue log to a database, it would be very easy. Total talk time on a queue: - Sum of parameter 2 of COMPLETECALLER and COMPLETEAGENT events. - Sum of parameter 4 of TRANSFER event if your calls are transferred somewhere else in the dialplan (which could be another queue, so you have the rest of the duration on the other queue call). And you could filter per agent, queue, date, etc... Just my two cents. > I also looked at using the ResetCDR() or ForkCDR() dialplan functions, > but I don't see a way to cause code to run immediatly after the agent > answers a call from the queue. > > Any suggestions? Am I missing some easy way of doing this? > > Thanks! > > Scott. > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center PBX: (+57 1)6500800 ext. 1201 Fax: (+57 1)6500816 Móvil: (+57)3138873587 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MOH
Dears
-How can I stop MOH when status of the dial is ringing and let the user hear
the Ring Back Tone from the termination Gateway.
Even I can see in the CLI debugging the status is ringing
-my idea is to add music on hold stop when asterisk detect --
SIP/OPNS-096456c0 is ringing line
In which script this line located?
-- Executing [97130245...@default:1]
SetMusicOnHold("SIP/xx.xx.xx.xx-096ca8c0", "English") in new stack
-- Executing [9713024...@default:2] Dial("SIP/xx.xx.xx.xx-096ca8c0",
"SIP/OPNS/9713024561|300|m") in new stack
-- Called OPNS/9713024561
-- Started music on hold, class 'English', on SIP/xx.xx.xx.xx-096ca8c0
-- SIP/OPNS-096456c0 is ringing
-More over when I used directrtpsetup=yes I heard the MOH and the ring back
tone together .
Regards
*
No employee or agent is authorized to conclude any binding agreement on behalf
of Xplorium with another party by e-mail without express written confirmation
by an officer of Xplorium. Any views expressed by an individual in this
electronic message do not necessarily reflect views of Xplorium or its
subsidiaries and associates.
This electronic message and its attachments are solely addressed to the
addressee(s), and contain confidential information protected from disclosure
belonging to Xplorium.
If you are not the intended addressee of this electronic message and its
attachments, kindly delete it immediately from your system and notify the
sender by electronic mail. You must not copy this message or attachment or
disclose its content to any other person.
Xplorium does not guarantee the integrity of this electronic message and any of
its attachments, or that they are free from computer viruses or other defects.
*
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] T38modem in loopback mode does not work on asterisk 1.4.20.1
On Tue, Apr 14, 2009 at 9:52 AM, Florian Hackenberger wrote: > With asterisk 1.6, is it possible to use hylafax, or would asterisk > terminate the fax calls itself? With app_fax integrated into asterisk-1.6, you have an 'infinite' modem pool that you control through the dial-plan. Using dialplan variables you provide a filename to save the fax to, and you can use other dialplan directives to describe what you want to do with the file once you have it. One call gets one 'modem' that is dynamically allocated, then destroyed when the call completes. This dialplan auto-scales to the load capacity of your system / circuits. With app_fax there's no need to pre-allocate and control the 'modem' with another piece of software. It may be possible to use hylafax, but I don't know how or why you would. I thought the whole point of hylafax is to control a "modem", and Hylafax would tie you to a specific number of modems you would have to leave idling. My hunch is you can't do this with app_fax and 1.6, and would instead have to use one of those plug-ins you used to have to use with 1.4. Unless of course I'm misunderstanding something fundamental to hylafax. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] .GSM -> .WAV (or ,MP3) Conversion
On Tue, 14 Apr 2009, Arjan Kroon | Mobillion wrote: > I record the message in ULAW > > After that I call sox with this command: > /usr/bin/sox -c 1 -1 -t ul -r 8000 $in_fl -t wav -2 -r 8000 -c 1 > $wav_fl What are "-1" and "-2" for? Both sox 12.17.5 and 12.18.1 say they are invalid. Thanks in advance, Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] T38modem in loopback mode does not work on asterisk 1.4.20.1
On Tuesday 14 April 2009, Michael wrote: > > asterisk-1.6 with app_fax built-in > > Try 1.6. You'll be glad you did. > While I have not tried Asterisk 1.6 because I settled on Callweaver > at the time (which has native T38 support), I *strongly* recommend > going with software that has native T38 support. > > This could be Asterisk 1.6 or Callweaver. > > So +1 for the above. Thank you all for your comments. I'm unfortunately stuck with asterisk 1.4, because it took a considerable amount of time to patch it to be stable (and feature complete) enough for my requirements. Integrating Callweaver might be an option, however I would loose the advantage of easily controlling the fax call before handing it off to hylafax. With asterisk 1.6, is it possible to use hylafax, or would asterisk terminate the fax calls itself? Are there any success stories with t38modem, asterisk and hylafax? Cheers, Florian -- DI Florian Hackenberger flor...@hackenberger.at www.hackenberger.at ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ignoring time spent waiting in queue in CDR
On Tue, Apr 14, 2009 at 4:15 PM, Jared Smith wrote: > - "Scott Gifford" wrote: >> The CDR information contains the entire >> duration of the call as billable seconds, including time spent >> waiting >> in the queue. I would like the billable seconds to only include the >> time spent actually talking to an agent. > > You're absolutely right -- the CDR information is for the entire call. > Instead, look at the queue log (typically written to > /var/log/asterisk/queue_log). It will tell you most (if not all) of the > information you need for creating call queue reports. > Most, but not all.. Short answer - do an ResetCDR() before entering Queue. This will set CDR Answer status to "NO ANSWER", and next answer by agent will answer the CDR, so You will have two distinct values - duration and billsec. Duration will be total length, but billsec will be conversation time. On the other hand, You can easily link queue_log with CDR, by enabling storing of UNIQUEID within CDR record. The same UNIQUEID will be in queue_log for CONNECT and HANGUP events. We do have purely CDR based billing implemented, but it requires some attention upon upgrading Asterisk, as some tiny details might change, so careful testing is a must. We are happy, as it allows to see complete call flow for every call, group them easily etc. There's a sample screenshot: http://ftp.iq-labs.net/screenshots/cdr_view.jpg However You should really have a think about what are Your requirements, and how they could change in future. Perhaps using the queue_log would allow rapid implementation and changes. Also, make sure to take a look at queue_log on Asterisk 1.6.0/1.6.1, they have some nice features added. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs Inc, a...@iq-labs.net Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP and FW settings
Michael wrote: > On Tue, 14 Apr 2009 20:47:29 you wrote: > >> Hi michael, >> >> you should open both tcp,udp 5060,5061 too and as you mentioned between >> 1-2. >> > > AFAIK 5061 TCP is for TLS SIP which isn't used much yet? > > Is TCP the default for 5060, with UDP as fallback, or is this provider > dependent? > > Michael > > That's provider dependent. MOST SIP is done with UDP. Some people use TCP to get past firewalls or try and alleviate NAT issues, but it's non-standard and falls into the category of 'complete hack' where SIP is concerned. TCP is allowed via the RFC, I believe (I vaguely remember a transport=tcp setting somewhere in a header field), but whether or not it's supported by the provider or software varies widely. Microsoft uses only TCP in their communicator product, I think. Some clients will let you choose TCP or UDP. But for the most part, when dealing with a default asterisk install and your own phones/softphones, you shouldn't need to worry about TCP. N. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trunks
Turn off the 'qualify' setting on the peer(s) in sip.conf to stop it sending OPTIONS pings. As for it responding with a 200 OK to such, there is no way to turn that off. -- Sent from mobile device On Apr 14, 2009, at 8:14 AM, "Khaled W. Chehab" wrote: > Dears > > How to disallow asterisk to send the keep alive 200 ok message to > the peers > and trunks. > > > Regards > > > > > * > No employee or agent is authorized to conclude any binding agreement > on behalf of Xplorium with another party by e-mail without express > written confirmation by an officer of Xplorium. Any views expressed > by an individual in this electronic message do not necessarily > reflect views of Xplorium or its subsidiaries and associates. > > This electronic message and its attachments are solely addressed to > the addressee(s), and contain confidential information protected > from disclosure belonging to Xplorium. > > If you are not the intended addressee of this electronic message and > its attachments, kindly delete it immediately from your system and > notify the sender by electronic mail. You must not copy this message > or attachment or disclose its content to any other person. > > Xplorium does not guarantee the integrity of this electronic message > and any of its attachments, or that they are free from computer > viruses or other defects. > * > > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ignoring time spent waiting in queue in CDR
- "Scott Gifford" wrote: > The CDR information contains the entire > duration of the call as billable seconds, including time spent > waiting > in the queue. I would like the billable seconds to only include the > time spent actually talking to an agent. You're absolutely right -- the CDR information is for the entire call. Instead, look at the queue log (typically written to /var/log/asterisk/queue_log). It will tell you most (if not all) of the information you need for creating call queue reports. --- Jared Smith Training Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] T38modem in loopback mode does not work on asterisk 1.4.20.1
On Wed, 15 Apr 2009 00:43:45 you wrote: > Now for the part I do know something about. Native asterisk fax > support and native asterisk sip support improved in 1.6. With 1.6 > there is a built-in app_fax module which works quite well for sending > fax over SIP with T.38. I found the configuration and debugging > simpler and more understandable. I never really knew why my > asterisk-1.4 faxing experiments went so badly, and I had no reason to > go find out once my asterisk-1.6 faxing worked so well. > > I will say as an opinion: > Openh323 + ptlib_unix + t38modem + hylafax + asterisk-1.4 > > sounds like a lot harder to troubleshoot than: It's hell (from experience) > asterisk-1.6 with app_fax built-in > > Try 1.6. You'll be glad you did. While I have not tried Asterisk 1.6 because I settled on Callweaver at the time (which has native T38 support), I *strongly* recommend going with software that has native T38 support. This could be Asterisk 1.6 or Callweaver. So +1 for the above. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] .GSM -> .WAV (or ,MP3) Conversion
On Wed, 15 Apr 2009 00:30:43 David Backeberg wrote: > Once it's a wav you can mp3 it with lame or your preferred encoder, > but encoding and playing mp3s takes more cpu than just playing it in > gsm, or stopping after sox and playing as a wav. > > > Has anyone got any suggestions based on previous experience? I convert to MP3 and delete the original wav file using a script automatically executed by the Monitor command (because wav is very large files). ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] T38modem in loopback mode does not work on asterisk 1.4.20.1
On Tue, Apr 14, 2009 at 7:11 AM, Michael wrote: > On Tue, 14 Apr 2009 23:00:02 Florian Hackenberger wrote: > >> Can somone spot the problem? Is someone using t38modem with asterisk >> successfully? > The best advice I can offer is to give up now and use Callweaver otherwise you > can spend hours, or days, with no working result. I'll be a little more constructive than that. I've never done it the way you're trying to do it, and don't know whether it will work. I will say that I dis-recommend trying to test faxing, or even SIP in loopback mode. My experience is that debugging is more fruitful when you have two systems you can control, and this applies specifically to faxing where you want to be able to watch the sender and receiver separately. If you can disentangle the logs in real-time, more power to you. Fire up a virt if you don't have a second machine you can control. Now for the part I do know something about. Native asterisk fax support and native asterisk sip support improved in 1.6. With 1.6 there is a built-in app_fax module which works quite well for sending fax over SIP with T.38. I found the configuration and debugging simpler and more understandable. I never really knew why my asterisk-1.4 faxing experiments went so badly, and I had no reason to go find out once my asterisk-1.6 faxing worked so well. I will say as an opinion: Openh323 + ptlib_unix + t38modem + hylafax + asterisk-1.4 sounds like a lot harder to troubleshoot than: asterisk-1.6 with app_fax built-in Try 1.6. You'll be glad you did. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ignoring time spent waiting in queue in CDR
My suggestion is to use a tool made specifically for this - we happen to sell one, but there are many options with different prices and licencing model. Don't reinvent the wheel and concentrate on added value. l. 2009/4/14 Scott Gifford > Hello, > > I'm working on an Asterisk configuration for a call center, and they > bill based on the time spent talking to an agent, but not for any time > spent waiting in a queue. The CDR information contains the entire > duration of the call as billable seconds, including time spent waiting > in the queue. I would like the billable seconds to only include the > time spent actually talking to an agent. > > I am using Asterisk 1.4.18. > > The only way I have found so far is to correlate the CDRs with the > "CONNECT" queue records, figure out the end time of the call by adding > the CDR start time to the duration, then figure out the actual > duration by subtracting the time of the queue "CONNECT" record. That > seems messy and error-prone, and I'm hoping there's a better way. > > I also looked at using the ResetCDR() or ForkCDR() dialplan functions, > but I don't see a way to cause code to run immediatly after the agent > answers a call from the queue. > > Any suggestions? Am I missing some easy way of doing this? > > Thanks! > > Scott. > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- Loway - home of QueueMetrics - http://queuemetrics.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] .GSM -> .WAV (or ,MP3) Conversion
Hey,
I record the message in ULAW
exten => s,1,Record(${A_record}:ulaw,0,60)
After that I call sox with this command:
/usr/bin/sox -c 1 -1 -t ul -r 8000 $in_fl -t wav -2 -r 8000 -c 1
$wav_fl
Regards,
Arjan Kroon
Mobillion BV
-Oorspronkelijk bericht-
Van: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] Namens Tim Dobson
Verzonden: 14-04-2009 13:39
Aan: asterisk-users@lists.digium.com
Onderwerp: [asterisk-users] .GSM -> .WAV (or ,MP3) Conversion
Hey there,
I'm trying to convert some call recordings from asterisk we have in .gsm
format to something I can pipe through ffmpeg - wav would be good, mp3
would be amazing!
I've been trying playing with sox but I don't seem to be getting too far
with
1239101491.30.gsm -ql -r 64000 -t wav 1239101491.30.conv.wav resample
as ffmpeg borks at it:
t...@freee-meee:~/dmc/call recordings$ ffmpeg -i 1239101491.30.conv.wav
1239101491.30.conv.mp3
FFmpeg version r11872+debian_3:0.svn20080206-12ubuntu3.1, Copyright (c)
2000-2008 Fabrice Bellard, et al.
configuration: --enable-gpl --enable-pp --enable-swscaler
--enable-x11grab --prefix=/usr --enable-libgsm --enable-libtheora
--enable-libvorbis --enable-pthreads --disable-strip --enable-libfaad
--enable-libfaadbin --enable-liba52 --enable-liba52bin
--enable-libdc1394 --disable-armv5te --disable-armv6 --disable-altivec
--disable-vis --enable-shared --disable-static
libavutil version: 49.6.0
libavcodec version: 51.50.0
libavformat version: 52.7.0
libavdevice version: 52.0.0
built on Mar 13 2009 17:48:10, gcc: 4.3.2
Input #0, wav, from '1239101491.30.conv.wav':
Duration: 00:00:06.7, bitrate: 1040 kb/s
Stream #0.0: Audio: libgsm_ms, 64 Hz, mono, 1040 kb/s
File '1239101491.30.conv.mp3' already exists. Overwrite ? [y/N] y
Output #0, mp2, to '1239101491.30.conv.mp3':
Stream #0.0: Audio: mp2, 64 Hz, mono, 64 kb/s
Stream mapping:
Stream #0.0 -> #0.0
[mp2 @ 0xb7d352f0]Sampling rate 64 is not allowed in mp2
Error while opening codec for output stream #0.0 - maybe incorrect
parameters such as bit_rate, rate, width or height
t...@freee-meee:~/dmc/call recordings$
Has anyone got any suggestions based on previous experience?
www.tdobson.net
If each of us have one object, and we exchange them, then each of us
still has one object.
If each of us have one idea, and we exchange them, then each of us now
has two ideas. - George Bernard Shaw
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
