Re: [Asterisk-Users] my asterisk crashed
On 5/3/06, Goke Aruna [EMAIL PROTECTED] wrote: ... #0 ast_var_name (var=0x1) at chanvars.c:71 #1 0x0808934e in pbx_builtin_getvar_helper (chan=0x0, name=0xf5bc2d46 OUTBOUND_GROUP) at pbx.c:5904 #2 0xf5bbe1e4 in dial_exec_full (chan=0xa281820, data=0x0, peerflags=0xf469fee8) at app_dial.c:964 #3 0xf5bc23ed in dial_exec (chan=0x0, data=0x1) at app_dial.c:1601 This indicates a corrupted global variable list, This issue was fixed in 1.2.6, please upgrade. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Error : ast_readaudio_callback: Failed to write frame
On 4/30/06, Hatami Nugraha [EMAIL PROTECTED] wrote: Hi all, I always get this error message after I hangup a call, what does it mean ? WARNING[8957]: file.c:583 ast_readaudio_callback: Failed to write frame This means you hungup while asterisk was trying to play a file to you. It should be of no concern as long as it does not happen during a call. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Speeding up the dial of DTMF's in SIP channel
Please Ignore if you cannot edit the code.
You will have to modify app_dial.c in apps directory.
Look for code that calls ast_dtmf_stream(chan, ..., timeout)
The last parameter is the inter digit timeout, it can be set to as low
as 1 (1 millisec) a value of 0 it will default to 100millisecs.
The solution is to add an option to dial application for the timeout
which defaults to the current value(250ms) in app_dial which will
provide for custom timeouts through the dialplan.
Also note that too small timeouts like below 100ms will mess up inband
dtmf tones for example in some zap channels.
Imran
On 3/15/06, Álvaro Palma [EMAIL PROTECTED] wrote:
I'm dialing DTMF's in a SIP channel using the options:
[sip.conf]
dmtfmode=info
[extensions.conf]
exten = _XXX,1,Dial(SIP/gateway,,D(${EXTEN}))
(this is a custom SIP gateway, which receives the DTMF's sent from
softphones through Asterisk, and based on them, build the destination
PSTN number).
My problem is that Dial send the DTMF's to the SIP/gateway user at a
rate of about 1 DTMF each 300ms. I'd like to know if it's possible to
speed up that rate, or even, if it's possible to send the entire
extension as a single DTMF string.
Does anybody has a clue about how to do this? I was looking the options
for the Dial command, and nothing like that appears on it.
Thanks a lot for your help.
--
Atly.
Álvaro Palma
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Re: [Asterisk-Users] TE411P VPM
Use: modprobe wct4xxp vpmsupport=0 On 3/1/06, Aaron Daniel [EMAIL PROTECTED] wrote: Does anyone know how to disable the VPM in software rather than removing the card altogether? The canceler isn't working as well as the software cancelers were. Aaron ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: RE : [Asterisk-Users] lists problem, Gmail????????
I have experienced similar problems using gmail. Gmail certainly had some problems with emails from asterisk lists. I donot know if it was only restricted to asterisk lists. As not all emails were being delayed (or dropped), some of you might be under the impression that theres no problem. Please compare your emails with the list archives to be sure you didnt miss something important. Also, the problems seems to have gone away this week. Regards Imran On 2/13/06, Joseph Tanner [EMAIL PROTECTED] wrote: May be some truth to it though :( Personally I use gmail, but use a different email address that is forwarded to my gmail account. With this setup, I haven't had any issues. I use gmail because it's easily accessible from any PC, and I like how it groups conversations (probably why you see a lot of gmail addresses signed up on mailing lists). Joseph Tanner On 2/13/06, Olivier.taylor [EMAIL PROTECTED] wrote: Pfff, What for an answer :( I use gmail and have no problems. Olivier -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Martin Joseph Envoyé : lundi 13 février 2006 20:36 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [Asterisk-Users] lists problem, Gmail On Feb 13, 2006, at 2:45 AM, Simone Cittadini wrote: C F ha scritto: Am I the only one having trouble with this list? Since the begining of the week I have not been receiving mail from the list like I used to, is this a gmail problem? or is it subscription problem? or is something wrong with the list? anybody else using gmail having any problems? Yes, I'm also getting some lag sometimes, one or two days without receiving mails get a real mail server and it works great! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] echo cancel from telco
here is a little explanation: End user (You) - Your Telco -- Carrier 1 --- Carrier 2 Carrier 3 --- Carrier 4(PTT) --- Far End User So basically, the Echo cancelling work backwards usually cancellation for you would be done by Carrier 4, 3, 2, 1, or your Telco in that order and echo for the Far End User would be done by Your Telco, Carrier 1, 2, 3, or 4 in that order. Why in that order? AFAIK, the order is exactly the opposite, and if the user is experiencing echo on the sip phone, its most likely that the other end is the source of echo, which should be cancelled by the telco because its is nearer to the source of echo than the sip phone gateway. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] echo cancel from telco
On 2/7/06, Imran Ahmed [EMAIL PROTECTED] wrote: here is a little explanation: End user (You) - Your Telco -- Carrier 1 --- Carrier 2 Carrier 3 --- Carrier 4(PTT) --- Far End User So basically, the Echo cancelling work backwards usually cancellation for you would be done by Carrier 4, 3, 2, 1, or your Telco in that order and echo for the Far End User would be done by Your Telco, Carrier 1, 2, 3, or 4 in that order. Why in that order? AFAIK, the order is exactly the opposite, and if the user is experiencing echo on the sip phone, its most likely that the other end is the source of echo, which should be cancelled by the telco because its is nearer to the source of echo than the sip phone gateway. Never mind! I took the wording in a wrong way. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Rtp packets being dropped
AFAIK asterisk does not drop the packets, it just turns them into silence if it detects a dtmf. On 2/6/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hello Friends, I am experiencing a problem. The rtp packets which detect dtmf from inband are being dropped. I have tried a priority ip address which allows voip packets first but it didnt work out. Asterisk is dropping only dtmf packets. I am using Sip protocol. Is there any way in asterisk whereby I can detect the dropped packets or enable their queueing or buffering? Please help, I am running out of ideas. Thanking you all. With warm regards. Vivek J. Joshi. [EMAIL PROTECTED] Trikon electronics Pvt. Ltd. --Sweat saves blood, blood saves lives, and brains saves both. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] meetme and dtmf
Step 3 The Iax client heve to send some other DTMF to the IVR. How is the IVR still involved if the call has been transferred into a conference room? The IVR records the conversation between the other partecipant to the conference and wait '#' to stop recording and a '1' to save the file. may or may not work, try at your own risk: 1) Use a sip soft phone and set the dtmf mode = inband. 2) In asterisk set the dtmf mode for that soft phone to be rfc2833 or info. (this is done so that asterisk ignores the inband dtmf on the sip channel). 3) Design your dialplan such that asterisk should not depend on dtmf from the sip call. ex: exten xxx, 1, dial(zap/g/client_number) //on answer directed to conference room exten xxx, 2, dial(zap/g/ivr_number) //on answer directed to conference room. exten xxx, 3, meetme(conference room) once the sip call is in the conference then the ivr will detect dtmf from the audio data. Note that before the sip call is in a conference dtmf will not be detectable by the ivr or asterisk, and Ofcourse, this is not tested and only a test can confirm if it works. drawbacks: dtmf will not be available to ivr until your call is in conference. asterisk will never see any dtmf (which should be okay in this specific case). dtmf tones are not squelched so the other user in the conference will hear dtmf tones. Imran ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] meetme and dtmf
AFAIK there's no DTMF option in IAX2... IAX always sends DTMF inline, eliminating the confusion often found with SIP. http://www.voip-info.org/wiki-IAX Even though no IAX client supports inband dtmf, An IAX client can send inband dtmf which would have corrected your problem. The problem here is with the meetme application when dealing with non zaptel channels it does not have a mechanism to enable dtmf to pass through the conference unless dtmf is inband (i.e. part of the audio stream). The following are the solutions a) Use a SIP phone with inband dtmf (No guarantee this will work either) b) Modify meetme to broadcast dtmf to all channels in conference( All channels will work in this case). Imran ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] meetme and dtmf
On 2/1/06, Kevin P. Fleming [EMAIL PROTECTED] wrote: Imran Ahmed wrote: Even though no IAX client supports inband dtmf, An IAX client can send inband dtmf which would have corrected your problem. No, it won't. No IAX2 client will start a DSP to listen for inband DTMF, because IAX2 is defined to always send out-of-band DTMF. At best, if the receiving IAX2 system is just passing the audio along to another protocol that does support inband DTMF, then sending it in the audio stream would work. If the application receiving the DTMF is on the other IAX2 end, though (like MeetMe in this case), then it will never 'see' the DTMF, because Asterisk will not look in the audio stream for DTMF. I agree, but the other ends of the conference were zap channels in this case, at least that is what I figured by the first email. Imran. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] meetme and dtmf
Here is my problem, at this point the IVR doesn't hear the dtmf sended by the iax client, even if it can hear the dtmf sended by the first zap channel. I donot know if IaxComm has inband dtmf mode available, if so enable it and see if it works. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Pri Gateway Hardware
You donot need multiple asterisk boxes for a single t1. A single p4 box should be helpful, you can use digiums te110p pci card for a single pri line into the box. The same box could also be on another network dealing with SIP. On 1/9/06, Carlos Alperin [EMAIL PROTECTED] wrote: All that you need is at least two boxes: 1 is going to be the PRI Asterisk box, which interfaces with the outside world. Also has to be able to communicate via SIP or IAX with the second box. 2 is the real Asterisk pbx with all the extensions, and the pbx features. Also has to be able to communicate via SIP or IAX with the first box. Done. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Johnathan Falk Sent: Monday, January 09, 2006 4:20 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Pri Gateway Hardware Does anyone have any experience using a PRI gateway, I am looking for a way to have multiple asterisk boxes use one PRI, and send that over the network. I herd there are copper gateway devices (like a X100P card, only it registers with asterisk using sip, and it doesn't have to be physically connected to the box) Does anyone have any experience with a PRI gateway? And could tell me the cost and the quality? Thanks Johnathan Falk Network Administrator Clinton Community Schools ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk 1.2.1 and mixmonitor problem
I think the broken pipe issue is related with the mpg123 player, try disabling moh and see if it behaves the same way On 12/19/05, Maximiliano J. Goldsmid [EMAIL PROTECTED] wrote: I have the same problem !! :-( 2005/12/18, Mohammad Shokuie [EMAIL PROTECTED]: Hi there, Any one confronted a crash in asterisk when using mixmonitor app. When i'm using the mixmonitor app on a briged call as soon as the called party hangs up the call asterisk crashes and the process terminates with following error message : Segmentation fault. Ouch .. error while writing audion data :: broken pipe but when the calling party hangs up, everything is smooth. Anyone has any idea on this issue? TIA. M. Shokuie Nia _ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] number of users in a meetme conference
Hi All, I want to know what is the maximum number of users allowed in a single meetme conference. How far is this number practically feasible Thanks Imran ___ --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
