[Asterisk-Users] problems with 1.2 Beta1
Greetings!
I am running a small callcenter with 10
analog lines, aprox. 15 agents and using Asterisk 1.2beta1. We have 10
sipura 3000s connected to the PSTN and a few linksys PAP2s.
The ports connected to phones are configured as
SIP/200s and SIP/300s and the ones connected to the PSTN as
SIP/900s.
When an agents makes a call, asterisks bridges a
SIP/200 with a SIP/900. However, every now and then I see calls bridges between
two SIP/900s which of course should not occur. The agents claim then that
sometimes when they are on a call other agents can sneak in the
call.
Previously, when I was using version 1.0.9 and had
a similar problem which I fixed it with SetGroup and CheckGroup. When I upgraded
to 1.2Beta1 I replaced those two funtions with the corresponding functions in
the new version, but it appears these two functions don't work as they used to,
and that's why the lines are getting mixed. My extensions.conf looks
like:
[macro-stdial]
exten =>
s,1,NoOp(${GROUP_COUNT(L_${ARG1})})
exten => s,2,Set(GROUP()=L_${ARG1})
exten =>
s,3,NoOp(${GROUP_COUNT(L_${ARG1})})
exten =>
s,4,GotoIF($[${GROUP_COUNT(L_${ARG1})}>1]?${EXTEN}|106:${EXTEN}|5)
exten => s,5,Dial(SIP/${ARG1}/${ARG2},45,grTH)
exten => s,6,AGI(calif.agi)
exten => s,7,hangup
exten => s,106,NoOP
The agents also claim that the calls sometimes hangup abruptly while they
are on the phone. I don't have more info than that, other than this occurs on
just any ATA device. Any ideas on how can i debug these problems?
Thanks much
Dan
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[Asterisk-Users] chanisavail...not workin with SIP and IAX
all I cannot get ChanIsAvail to work with sip or iax on v1.0.3. It does work fine on a zap channel. I am trying with Sipuras and PAP2s. It appears I am not the only one having this problem. Has anyone gotten it to work? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Service Unavailble, Sipura 3000, CheckGroup, what the heck??
Folks! I discovered some serious problem with several Sipuras 3000 but I don't know if the problem is with them or Asterisk. Basically, if I call a Sipura PSTN line, when there is a call already in progress, generally I get a 503 Sevice Unavailable, but if I try hard enough, I am able to get through and connect to dialed number. The other call gets disconnected but the originator of the other call is now on my call. Is this a bug of Asterisk's SIP implementation? or is it a Sipura configuration problem? I looked at other alternatives but haven't had any luck. Hint didn't work and CheckGroup does exactly the same thing. Sometimes I get Service Unavailable but other times i can dial even though there is a call in progress. Any ideas? . ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] sipura3000 problems in callcenter
I have 4 sipuras 3000 in a small callcenter connected to the PSTN receiving calls and forwarding them to Asterisk and viceversa. In addtiion I have some x100s, linksys FXSs, etc Strange things are happening with the Sipura and Asterisk which I cannot seem to figure out. During off hours at the callcenter, when no one is placing calls, if I place or receive a call with any of the Sipura, everything seems fine (well almost since it doesn't detect a hangup from the receiving end). However, when the call center is in full operation, if I do sip show channels, I can see several instances of each of the sipura SIP FXO channels as if they were being used, but they are not, since i can still place calls through them. Another strange thing I've notice is that in many instances, the lastapp field of CDR does not match with the dst field, and as far as I know, they should. For example if the dst field was 5551212 the lastapp would be SIP/905/5551211. Any help would be greatly appreciated. Dan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 302 Moved temporarily problem / Sipura 3000
I can send calls from asterisk to a Sipura FXO interface (SIP/300) from any SIP phones including SIP/205 which is the Sipura 3000 FXS interface. The problem I have is when a call from the PSTN is sends to Asterisk. On extnesion conf I dial all the SIP clients I get a 302 Moved temporarily when it dials SIP/205, the FXS interface. I have read on the bug tracker that ther is a patch with a new app SIPredirect (or similar) would this work for my problem. Any other thoughts? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] agent logoff
I am using AgentCallbacklogin to logon agents. I am trying to avoid agents being logged in more than once in different extensions (is this a bug?) by passing the callerid to the AgentCallbacklogin funtcion as an option. The problem is that by doing this, agents are not asked for an extension and they cannot logoff (by pressing the #). Any ideas how can agents logoff? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] no mail sent on voice message
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Re: [Asterisk-Users] FREE (305) and (786) termination. Anyone interested?
Alejandro Why can't you use IAX? I'd love to test your termination. Saludos Daniel - Original Message - From: Alejandro Sosa To: [EMAIL PROTECTED] Sent: Tuesday, July 20, 2004 2:54 PM Subject: [Asterisk-Users] FREE (305) and (786) termination. Anyone interested? I have an Asterisk box with free local termination to area codes (305) and (786) [Miami area, US]. I want to configure it to accept incomming VoIP traffic (cant use IAX) and terminate calls over the PSTN network. I need help with the configuration and also some incoming traffic for testing purposes. Please contact me if you can help. Regards, Alejandro.
[Asterisk-Users] Problems with festival
I cannot get Festival to work with asterisk. I have the following: exten => 555,1,Answerexten => 555,2,Festival(mary has a little lamb)exten => 555,3,Hangup I get the following from asterisk: "Festival returned ER" and the festival logs shows the following: client(1) Fri Jul 16 15:35:54 2004 : disconnectedclient(2) Fri Jul 16 15:40:26 2004 : accepted from localhost Festival seems to be running fine. For example if I do: echo this is a test | --tts --language english it works just fine I'm starting festival from the script festival_server and the logs shows no errors. I had to rename the festival directory to festival-1.4.3 to apply the patch Any ideas what can the problem be?
Re: [Asterisk-Users] problems with analog interface to PBX
Yes, I've tried with SendDTMF, and it works, but if I do that, then * looses control of the call. That is, the call is transfered to the new extensions on the PBX but since * is not in the calll flow anymore, it doesn't know if on the other end they have ansered or not. - Original Message - From: "Steven Critchfield" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Tuesday, May 18, 2004 5:56 PM Subject: Re: [Asterisk-Users] problems with analog interface to PBX > On Tue, 2004-05-18 at 15:45, Dan Fernandez wrote: > > Steve, > > > > Thanks for your respnose. The flash does seem to work. If I plug a phone on > > the x100p I can hear with the x100p flashes. I then get a dialtone. The > > problem is that when i try to dial again from that card, i get "cannot > > create zap channel". It seems that because the line is now off hook, the > > dial cannot proceed. > > Without having read the thread, flash returns you to the channel. From > that point use senddtmf to "dial" the numbers you want on the channel > you already have. > > > - Original Message - > > From: "Steve Creel" <[EMAIL PROTECTED]> > > To: <[EMAIL PROTECTED]> > > Sent: Thursday, May 13, 2004 11:04 AM > > Subject: Re: [Asterisk-Users] problems with analog interface to PBX > > > > > > > On Wed, 12 May 2004, Dan Fernandez wrote: > > > > > > >Folks, > > > > > > > >For the last few days I've been trying to experiment with a Panasonic PBX > > > >and an X100P but have run into quite a few problems which I am not sure > > > >if they can be solved with this type of card (how about TDM01B?) > > > > > > > >1) I wanted to use *'s IVR capabilities, so I routed the calls to the > > > > extension where the x100p was connected to. > > > > > > > >Asterisk should answer the call, playback a message, dial another PBX > > > >extension and if no one answers dial another extension (via IAX). > > > > > > > >The first problem I ran into was that the Flash application doesn't > > > >really work. To get around this I added another x100p to dial the new > > > >extension. The problem I ran here was that even though I specified in the > > > >Dial app to just dial for 30 seconds, it rang forever as if * cannot > > > >recongnize that no one had picked up. Asterisk does seem to detect > > > >hangups and busy tones (I have busydetect=yes and busycount=10) > > > > > > For about 6 months, we were using the same logical setup (a channelbank of > > > FXO cards for a Merlin Legend switch, with asterisk doing incoming IVR / > > > autoattendant, then transferring the calls out to the Legend, and > > > handling voicemail). The first problem I encountered that I hadn't > > > expected had to do with asterisk transferring the call back to the Legend. > > > I did a Flash(), a SendDTMF(), and another Flash() - the Legend saw this > > > as an attended transfer, and it caused some oddities. Turns out I needed > > > to Flash(), SendDTMF(), Hangup(). Along the way, I found the Flash times > > > that the legend was expecting to see, and adjusted them in the source > > > code, so as to eliminate occasional flash detection problems. > > > > > > I'd take time to plug an analog set into the extension you have the X100P > > > on, and make sure you can flash/transfer calls like you're expecting > > > asterisk to. There's no reason (that I know of) that your flash can't > > > give you exactly the behavior you're looking for. > > > > > > Good luck to you, > > > > > > Steve > > > ___ > > > Asterisk-Users mailing list > > > [EMAIL PROTECTED] > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > To UNSUBSCRIBE or update options visit: > > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > -- > Steven Critchfield <[EMAIL PROTECTED]> > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] problems with analog interface to PBX
Thanks for the response. Have you try the new TDM FXO cards? Does call progress work with those? - Original Message - From: "Vic Cross" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Thursday, May 13, 2004 5:46 AM Subject: Re: [Asterisk-Users] problems with analog interface to PBX > On Wed, 12 May 2004, Dan Fernandez wrote: > > > Asterisk should answer the call, playback a message, dial another PBX > > extension and if no one answers dial another extension (via IAX). > > > > The first problem I ran into was that the Flash application doesn't > > really work. To get around this I added another x100p to dial the new > > extension. The problem I ran here was that even though I specified in > > the Dial app to just dial for 30 seconds, it rang forever as if * cannot > > recongnize that no one had picked up. Asterisk does seem to detect > > hangups and busy tones (I have busydetect=yes and busycount=10) > > In the absence of call progress detection settings, Zap analog channels > tell Dial() that they are Connected more-or-less as soon as they have > completed dialling (I see this on the display of my 7960: I see Proceeding > for a second or two, then Connected, when I dial through an X100P). So, > the timeout on your Dial() never gets triggered because the channel > reports a connected call almost straight away. > > To do what you want, you would need callprogress=yes -- as long as your > Panasonic PBX generates authentic US tones. busydetect will only detect > busy (!), not ringback or congestion or any of the other tones you would > need to make your application work the way you want -- call progress > detection tries to do this for you. > > The bad news is that even if your PBX generates US tones, reports are that > the detection is not too reliable. > > > Am I trying to do something that the x100p is not capable of? Would > > making changes to the indications.conf help at all? > > It's not that the X100P can't do the job, it's more that analogue lines > can't do the job :) Seriously, if your PBX generates US tones then give > callprogress=yes a try. From my reading of the code, the tones specified > in indications.conf are unrelated to the way the * DSP does call progress > detection (have a look at functions like ast_dsp_call_progress() in dsp.c > if you're really curious). > > > 2) I would also like to use * for voicemail. The user should be able to > > dial the extension where the x100p is connected and asterisk recognized > > the extension the user is dialing and request for the password? Is this > > possible? > > On an analogue channel via an X100P, there is no "called number" > indication. So you can't tell what number the caller dialled to reach > you. If you wanted to use the * box as a voicemail-only machine, you > could drop the caller straight into VoiceMailMain, but if you wanted other > functions (conference rooms, VoIP gateway, etc) you would need to use an > IVR... > >"press 1 to access Voicemail... > press 2 to reach a Voice-over-IP user... > press 3 to join a conference... > ..." > > This doesn't really help your original need: to dial another number on the > PBX and get control back if needed. If callprogress=yes doesn't work for > you, you could try something like this (off the top of my head): > > exten => 4,1,Playback(trying-press-*-to-come-back) > exten => 4,2,Dial(Zap/1/1234,,Hg) > exten => 4,3,Goto(103) > exten => 4,103,Playback(sorry-cant-reach) > exten => 4,104,Goto(menu,s,1) > > On the Dial(), the option H enables caller hangup using '*', and g says go > on in context when the destination channel hangs up. This would put your > caller in the driver seat and get them to do the tone detection for you ;) > > > Hope this helps, > Vic Cross > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] problems with analog interface to PBX
Steve, Thanks for your respnose. The flash does seem to work. If I plug a phone on the x100p I can hear with the x100p flashes. I then get a dialtone. The problem is that when i try to dial again from that card, i get "cannot create zap channel". It seems that because the line is now off hook, the dial cannot proceed. Thanks again. - Original Message - From: "Steve Creel" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Thursday, May 13, 2004 11:04 AM Subject: Re: [Asterisk-Users] problems with analog interface to PBX > On Wed, 12 May 2004, Dan Fernandez wrote: > > >Folks, > > > >For the last few days I've been trying to experiment with a Panasonic PBX > >and an X100P but have run into quite a few problems which I am not sure > >if they can be solved with this type of card (how about TDM01B?) > > > >1) I wanted to use *'s IVR capabilities, so I routed the calls to the > > extension where the x100p was connected to. > > > >Asterisk should answer the call, playback a message, dial another PBX > >extension and if no one answers dial another extension (via IAX). > > > >The first problem I ran into was that the Flash application doesn't > >really work. To get around this I added another x100p to dial the new > >extension. The problem I ran here was that even though I specified in the > >Dial app to just dial for 30 seconds, it rang forever as if * cannot > >recongnize that no one had picked up. Asterisk does seem to detect > >hangups and busy tones (I have busydetect=yes and busycount=10) > > For about 6 months, we were using the same logical setup (a channelbank of > FXO cards for a Merlin Legend switch, with asterisk doing incoming IVR / > autoattendant, then transferring the calls out to the Legend, and > handling voicemail). The first problem I encountered that I hadn't > expected had to do with asterisk transferring the call back to the Legend. > I did a Flash(), a SendDTMF(), and another Flash() - the Legend saw this > as an attended transfer, and it caused some oddities. Turns out I needed > to Flash(), SendDTMF(), Hangup(). Along the way, I found the Flash times > that the legend was expecting to see, and adjusted them in the source > code, so as to eliminate occasional flash detection problems. > > I'd take time to plug an analog set into the extension you have the X100P > on, and make sure you can flash/transfer calls like you're expecting > asterisk to. There's no reason (that I know of) that your flash can't > give you exactly the behavior you're looking for. > > Good luck to you, > > Steve > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] problems with analog interface to PBX
Folks, For the last few days I've been trying to experiment with a Panasonic PBX and an X100P but have run into quite a few problems which I am not sure if they can be solved with this type of card (how about TDM01B?) 1) I wanted to use *'s IVR capabilities, so I routed the calls to the extension where the x100p was connected to. Asterisk should answer the call, playback a message, dial another PBX extension and if no one answers dial another extension (via IAX). The first problem I ran into was that the Flash application doesn't really work. To get around this I added another x100p to dial the new extension. The problem I ran here was that even though I specified in the Dial app to just dial for 30 seconds, it rang forever as if * cannot recongnize that no one had picked up. Asterisk does seem to detect hangups and busy tones (I have busydetect=yes and busycount=10) Am I trying to do something that the x100p is not capable of? Would making changes to the indications.conf help at all? 2) I would also like to use * for voicemail. The user should be able to dial the extension where the x100p is connected and asterisk recognized the extension the user is dialing and request for the password? Is this possible? Thanks Dan
[Asterisk-Users] x100p / Answer-> Flash -> Dial
I have an X100P connected to an extension of a Panasonic PBX. When a call from the PSTN comes in, it is routed directly to the extension where the x100p is . I want * to answer the call, play a message and then transfer the call to another extension via the Zap channel where the call was received (I need to flash the zap channel) . If this extension doesn't answer I want then to dial an IAX channel. The problem is that when I do a Flash on the zap channel, and then try to dial a new extension via that zap channel I get the following error "can't create zap channel". If I do a SendDTMF() the call does get transfer to the new extension but then * gets out of the call loop and don't know it is answered or not by the new extension. Am I missing something? Why am I getting the "can't creat za channel" Thanks in advance. Dan
[Asterisk-Users] x100p / Answer-> Flash -> Dial
I have an X100P connected to an extension of a Panasonic PBX. When a call from the PSTN comes in, it is routed directly to the extension where the x100p is . I want * to answer the call, play a message and then transfer the call to another extension via the Zap channel where the call was received (I need to flash the zap channel) . If this extension doesn't answer I want then to dial an IAX channel. The problem is that when I do a Flash on the zap channel, and then try to dial a new extension via that zap channel I get the following error "can't create zap channel". If I do a SendDTMF() the call does get transfer to the new extension but then * gets out of the call loop and don't know it is answered or not by the new extension. Am I missing something? Why am I getting the "can't creat za channel" Thanks in advance. Dan
Re: [Asterisk-Users] FIXED : cdr_addon_mysql problem linking
I finally figured it out. Had to install zlib-devel package. sorry for the posting, but it was driving me nuts. - Original Message - From: Dan Fernandez To: [EMAIL PROTECTED] ; [EMAIL PROTECTED] Sent: Monday, February 23, 2004 8:07 PM Subject: [Asterisk-Users] cdr_addon_mysql problem linking I have Suse 9.0 with gcc3.3.1 (didn't have any problem with the previous version of gcc )and when I run make install I get the following error: /usr/lib/gcc-lib/i586-suse-linux-/3.3.1/../../../.../i586-suse-linux/bin/ld: cannot find -lz Any help would be appreciated. Dan
[Asterisk-Users] cdr_addon_mysql problem linking
I have Suse 9.0 with gcc3.3.1 (didn't have any problem with the previous version of gcc )and when I run make install I get the following error: /usr/lib/gcc-lib/i586-suse-linux-/3.3.1/../../../.../i586-suse-linux/bin/ld: cannot find -lz Any help would be appreciated. Dan
[Asterisk-Users] RxFax
I am also having problems receiving my first fax. I get a 320byte file (for a4 page fax). If I look a the tiff generated, is just has some few dots.I am sending the fax from a notebook with Windows XP to an X100P and usinglibtiff v3.5.7.Has anyone successfully received faxes ?Output to the console as follows:Changed from phase 0 to 1> Start receiving document> Changed from phase 1 to 4> Sending ident> >>> CSI: 40 38 37 36 35 34 33 32 31 20 20 20 20 20 20 20 20 20 20 20> 20> DIS:> Store and forward Internet fax: no> Real-time Internet fax: no> Preferred octets: 256> Can receive fax> Data signalling rate: V.29> R8x7.7lines/mm and/or 200x200pels/25.4mm OK> 2D coding OK> Scan line length: 215mm> Recording length: A4 (297mm)> Receiver's minimum scan line time: 0ms at 3.85 l/mm: T7.7 = T3.85> R8x15.4lines/mm OK> Inch-based resolution preferred: no> Metric-based resolution preferred: no> Minimum scan line time for higher resolutions: T15.4 = T7.7> >>> DIS: 80 00 c6 f0 80 80 01> HDLC underflow in state 9> Changed from phase 4 to 3> <<< TSI: 43 20 20 20 20 20 37 39 36 39 36 33 32 20 31 31 39 20 39 34> 2b> TSI without final frame tag> Remote fax gave TSI as: "+49 911 2369697 "> <<< DCS: 83 00 46 f0 00> DCS with final frame tag> In state 9> DCS:> Store and forward Internet fax: no> Real-time Internet fax: no> Can receive fax> Data signalling rate: V.29, 9600bps> R8x7.7lines/mm and/or 200x200pels/25.4mm OK> Scan line length: 215mm> Recording length: A4 (297mm)> Minimum scan line time: 0ms> Get at V.29> Changed from phase 3 to 5> Fast carrier up> Fast carrier down> Changed from phase 5 to 4> 0 bad bits in trainability test> Start rx document - compression 1> Start rx page> >>> CFR: 84> HDLC underflow in state 5> Post trainability> Changed from phase 4 to 5> Fast carrier up> Fast carrier down> Fast carrier up> Fast carrier down> Fast carrier up> Equalizer state:> -7 ( 0.0, 0.0) -> 0.0> -6 ( 0.0, 0.0) -> 0.0> -5 ( 0.0, 0.0) -> 0.0> -4 ( 0.0, 0.0) -> 0.0> -3 ( 0.0, 0.0) -> 0.0> -2 ( -0.08332, -0.68161) -> 0.47154> -1 ( 0.66136, 0.74688) -> 0.99522> 0 ( 0.83336, 2.86588) -> 8.90777> 1 ( 0.66136, 0.74688) -> 0.99522> 2 ( -0.08332, -0.68161) -> 0.47154> 3 ( 0.0, 0.0) -> 0.0> 4 ( 0.0, 0.0) -> 0.0> 5 ( 0.0, 0.0) -> 0.0> 6 ( 0.0, 0.0) -> 0.0> 7 ( 0.0, 0.0) -> 0.0> Equalizer state:> -7 ( 0.24872, -0.01973) -> 0.06225> -6 ( 0.06295, -0.59891) -> 0.36265> -5 ( 0.04198, -0.41768) -> 0.17622> -4 ( 0.12922, 0.34088) -> 0.13290> -3 ( 0.29121, 0.41776) -> 0.25932> -2 ( 0.05779, -1.18501) -> 1.40758> -1 ( 0.67094, -0.35069) -> 0.57314> 0 ( 0.60670, 2.23125) -> 5.34655> 1 ( 0.34490, 1.36992) -> 1.99564> 2 ( -0.31233, 0.09159) -> 0.10594> 3 ( -0.00825, 0.00787) -> 0.00013> 4 ( 0.02226, -0.47177) -> 0.22306> 5 ( -0.17953, 0.14528) -> 0.05334> 6 ( -0.25318, 0.57800) -> 0.39819> 7 ( -0.10465, 0.05659) -> 0.01415> Equalizer state:> -7 ( 0.10674, 0.25606) -> 0.07696> -6 ( -0.05000, -0.10084) -> 0.01267> -5 ( -0.04246, -0.24755) -> 0.06308> -4 ( 0.01990, 0.30283) -> 0.09210> -3 ( -0.04673, 0.31177) -> 0.09939> -2 ( -0.20575, -0.84531) -> 0.75687> -1 ( 0.53295, 0.18096) -> 0.31678> 0 ( 0.84089, 2.50094) -> 6.96180> 1 ( 0.69048, 1.16134) -> 1.82547> 2 ( -0.06589, -0.16464) -> 0.03145> 3 ( -0.04671, 0.30535) -> 0.09542> 4 ( 0.07891, -0.09615) -> 0.01547> 5 ( 0.10829, 0.05362) -> 0.01460> 6 ( -0.01236, 0.00810) -> 0.00022> 7 ( -0.08830, -0.09249) -> 0.01635> Fast carrier training failed> Equalizer state:> -7 ( 0.08400, 0.24069) -> 0.06499> -6 ( -0.05179, -0.10933) -> 0.01464> -5 ( -0.02255, -0.26286) -> 0.06961> -4 (
[Asterisk-Users] pattern matching problem when dialing
I am having problems with early dialing and chan_phone. In extensions.conf I have: exten => _41.,1,Dial,IAX If I dial via a SIP or ZAP channels it works fine. With chan_phone it start dialing right after the 3rd number. If tried different combinations like (41., ... or _41X., ) and still the same problem. This used to work ok a few weeks back.!!
Re: [Asterisk-Users] Is there any MFC-R2 implementation for asterisk?
Any news on this regard? If this is not implemented yet, what alternatives do we have? A channel bank? - Original Message - From: "Paulo Mannheimer" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Thursday, September 11, 2003 10:23 AM Subject: RE: [Asterisk-Users] Is there any MFC-R2 implementation for asterisk? > Me too. I sent Steve an email about this, but didn't get a reply. > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of LQ > (Asterisk) > Sent: September 11, 2003 10:19 AM > To: [EMAIL PROTECTED] > Cc: [EMAIL PROTECTED] > Subject: [Asterisk-Users] Is there any MFC-R2 implementation for > asterisk? > > > > The last thing that I read about it was: > > Steve Underwood [EMAIL PROTECTED] wrote on Sep 3: > >> Is E&M designed to work with the E1 driver code? I think probably > >> not. I had to fix some things to get proper access to the CAS > >> signaling bits when I implemented MFC/R2... > So, apparently he implemented it. > I was trying to contact Steve, but he isn't answering me. > > Does anybody have any news about it? > > Regards, > Pablo. > > >> -Original Message- > >> From: Herry Sitepu [mailto:[EMAIL PROTECTED] > >> Posted At: Thursday, September 11, 2003 5:07 > >> Posted To: Asterisk > >> Conversation: [Asterisk-Users] Is there any MFC-R2 implementation for > > >> asterisk? > >> Subject: [Asterisk-Users] Is there any MFC-R2 implementation for > >> asterisk? > >> > >> > >> Hi guys, > >> Is there anyone has implemented MFC-R2 for astrisk? > >> > >> Regards > >> Herry Sitepu > >> > >> ___ > >> Asterisk-Users mailing list > >> [EMAIL PROTECTED] > >> http://lists.digium.com/mailman/listinfo/asterisk-users > >> > > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP segfault, problem loading modules, gdb output included
Last week I did a CVS update and since then I haven´t been able to run asterisk normally. The strange thing is that I have even go back to previous versions (0.5.0) and I am seening the same problems. Basically, when I try to load the zap module I get the following error: WARNING[16384]: File chan_zap.c, Line 577 (zt_open): Unable to> > specify channel 1: Device or resource busy (wxfxo loads fine) If I then do not load zap I get a problem when trying to load iax2 WARNING:Unable to bind to 0.0.0.0 port 4569: Address already> > in use If I then do not load iax2 asterisk starts fine. However when I try to place a SIP call it segfaults right away. The output of gdb asterisk /etc/asterisk/core.2906 follows: (gdb) bt #0 0x401507f1 in ?? () #1 0x444c7fab in ?? () #2 0x4002c020 in ?? () (gdb) Can someone please help me. Dan
[Asterisk-Users] SIP segfaults and problems loading modules
Last week I did a CVS update and since then I haven´t been able to run asterisk normally. The strange thing is that I have even go back to previous versions (0.5.0) and I am seening the same problems. Basically, when I try to load the zap module I get the following error: WARNING[16384]: File chan_zap.c, Line 577 (zt_open): Unable to> > specify channel 1: Device or resource busy (wxfxo loads fine) If I then do not load zap I get a problem when trying to load iax2 WARNING:Unable to bind to 0.0.0.0 port 4569: Address already> > in use If I then do not load iax2 asterisk starts fine. However when I try to place a SIP call it segfaults right away. I am starting asterisk as "asterisk -gc". I am also having trouble running gdb I get the following error: GDB was configured as "i586-suse-linux"... "/etc/asterisk/core.2035" not in executable fromat: File format not recognized. Can someone give me hand to get myself out of this mess? Thanks Dan
Re: [Asterisk-Users] problem loading chan_iax2.so and chan_zap.sofrom latest CVS
Steven Thanks for the help. After rebooting the box, * gives me an error claiming that * is already running on /var/run/asterisk.ctl Before I rebooted I ensured that there was no asterisk.pid or asterisk.ctl. After I get the above mentioned message if I then run asterisk again I get the "Unable to open...device busy" (with or without the asterisk.pid and/or asterisk.ctl) Any help would be greatly appreciated. Rgds Dan - Original Message - From: "Steven Critchfield" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Wednesday, September 17, 2003 12:02 AM Subject: Re: [Asterisk-Users] problem loading chan_iax2.so and chan_zap.sofrom latest CVS > On Tue, 2003-09-16 at 20:27, Dan Fernandez wrote: > > I just updated to the new CVS and now I am getting the following error > > from chan_zap (modprobe wcfxo works fine): > > > > WARNING[16384]: File chan_zap.c, Line 577 (zt_open): Unable to > > specify channel 1: Device or resource busy > > > > WARNING:Unable to bind to 0.0.0.0 port 4569: Address already > > in use > > This looks rather obvious to me that you may not have stopped the > previous asterisk install. Either that or you have a kernel problem and > (oddly) need to reboot to free the port and the device handles. > > -- > Steven Critchfield <[EMAIL PROTECTED]> > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] problem loading chan_iax2.so and chan_zap.so from latest CVS
I just updated to the new CVS and now I am getting the following error from chan_zap (modprobe wcfxo works fine): WARNING[16384]: File chan_zap.c, Line 577 (zt_open): Unable to specify channel 1: Device or resource busy ERROR[16384]: File chan_zap.c, Line 4781 (mkintf): Unable to open channel 1: Device or resource busy here = 0, tmp->channel = 0, channel = 1 ERROR[16384]: File chan_zap.c, Line 6498 (load_module): Unable to register channel '1' WARNING[16384]: File loader.c, Line 299 (ast_load_resource): chan_zap.so: load_module failed, returning -1 WARNING[16384]: File loader.c, Line 394 (load_modules): Loading module chan_zap.so failed! I did an lsmod and wcfxo is there. Also if I add noload chan_zap.so on modules.conf then it bombs when loading chan_iax2.so WARNING:Unable to bind to 0.0.0.0 port 4569: Address already in use WARNING ...chan_iax2.so: load_module failed, returning -1 Is anyone getting the same errors?
[Asterisk-Users] Re: [digium.com #860] Fw: x100P: Ring/off-hook in strange state 6 on channel1
Yes, setting "callprogress=no" fixed the problem. Thanks to everyone. - Original Message - From: "Martin Pycko via RT" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Tuesday, September 16, 2003 6:43 PM Subject: [digium.com #860] Fw: x100P: Ring/off-hook in strange state 6 on channel1 > it looks like if you turn off callprogress and busydetect that it helps. > Tell me if that helps in your case and we might modify the code so that > you could use callprogress/busydetect if you want to > > regards > Martin > > > [martinp - Fri Sep 05 11:19:08 2003]: > > > > I can try to log in to your box now and debug the callerid. I need the > > IP/password and the number of your PSTN line connected to X100P with all > > the prefixes to call from US. > > > > regards > > Martin > > > > > > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ATA v2.16 Register Update problem
I´ve got an ATA with v2.16 configured as per JTodd´s guide having problems re-registering. The only thing particular to this ATA (I´ve got several all configured the same way but one of them have problems) is that is on a public IP. I have commented NAT=1 and canreinvite=no on sip.conf The ATA registers fine the first time but on the following registration updates, * gives me the the warning: "Registration fail for ." and it continues trying to register unsuccesfully over and over. If I do sip show peers, the ATA is not there, of course. Looking at the archives I´ve found that on the new version (2.16.1) they have fixed a bug which is somewhat related: CSCeb17953 The Cisco ATA stops the registration process if it receives an unexpected response to a REGISTER request. HAs anyone had this problem before with 2.16? Woudl 2.16.1 fix this ? Thanks in advance
Re: [Asterisk-Users] PROBLEM RECIVING CALLS AT FXO
I´ve been having this same problem for a few weeks now. WARNING[262160]: File chan_zap.c, Line 2857 (zt_handle_event): Ring/Off-hook in strange state 6 on channel 1 I get this message and then the Zap channel hangs up and it does not Answer the call. I have no problems dialing out. This used to work just fine. - Original Message - From: "Adam Goryachev" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Thursday, September 11, 2003 9:50 PM Subject: RE: [Asterisk-Users] PROBLEM RECIVING CALLS AT FXO > This looks similar to a problem I had about 2 weeks ago, more details > below.. > > > I have the next problem.. I have a FXO card with i can make > > calls but i cant > > recive calls. > > I couldn't do either (reliably) > > > At the consol, i get the next error: > > > > -- Zap/2-1 is ringing > > -- Zap/2-1 is ringing > > -- Zap/2-1 answered Zap/1-1 > > -- Attempting native bridge of Zap/1-1 and Zap/2-1 > > WARNING[262160]: File chan_zap.c, Line 2857 (zt_handle_event): > > Ring/Off-hook > > in strange state 6 on channel 1 > > I had something similar to this, but also had other messages saying we had > received a ring even though we were off-hook. > > mark actually logged in and had a brief look, but I got tired and had to go > home, and haven't followed it up with him since. One of these days when I go > out on-site again, and can wait until 2am to try and catch Mark, I'll follow > it up, but it's kinda painful... > > Regards, > Adam > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] x100P: Ring/off-hook in strange state 6 on channel1
All of a sudden I am getting the following warning "Ring/off-hook in strange state 6 on channel1" from chan_zap.c and I cannot answer calls. I can place calls out without a problem though. Any ideas what can be the problem. I have checked zapata.conf and zaptel.conf and they both seem fine. Thanks in advance. Dan
[Asterisk-Users] problem with manager: Response error, Missing action in request
I am having problems using the manager even though I am following the instructions from the Manager.rtf doc. In manager.conf I have the following [general] enabled=yes port=5038 [fred] username=fred secret=fred read=system,call,log,verbose,command,agent write=system,call,log,verbose,command,agent I do the following: System prompt # telnet localhost 5038 Trying 127.0.0.1... Connected to localhost. Escape character is '^]'. Asterisk Call Manager/1.0 Action: Login Username:fred Secret: fred and the result I get is Response:Error Message:Missing action in request Any ideas what am i doing wrong?
Re: [Asterisk-Users] CDR-Event on AstManager
- Original Message - From: "Michiel Betel" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Monday, May 19, 2003 11:53 AM Subject: RE: [Asterisk-Users] CDR-Event on AstManager The manager inteface currently sends the following events with the associated parameters: Event: Newexten Channel Context Extension Priority Event: Newchannel Channel State Callerid Event: Hangup Channel Event: Rename Oldname Newname Event: Newcallerid Channel Callerid Event: Newstate Channel State Callerid Event: Link Channel1 Channel2 Event: Unlink Channel1 Channel2 Event: Hangup Channel Event: ExtensionStatus Exten Context Status Event: Reload Message: Reload Requested Event: Response Success XX Event: MessageWaiting Mailbox Waiting Event: Agentlogin Agent Channel Event: Agentlogoff Event: Join Channel Queue Position Event: Leave Channel Queue -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Thomas Haeger Sent: maandag 19 mei 2003 16:20 To: Asterisk User Subject: [Asterisk-Users] CDR-Event on AstManager Hi all, what's your opinion about CDR-Event (like Hangup or Ring etc.) on AstManager ? Or, is something like this already implemented ? Regards, Thomas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] iax2 and reinvites
Is there a way in iax to have to endpoints talk to each other directly (after the call is setup by *) without going through *. In sip, with * you can do it by configuring sip.conf with canreinvite = yes.
Re: [Asterisk-Users] RTP session traversing Asterisk server ...
On your sip.conf for each sip endopoint set canreinvite = yes. That way the rtp stream won´t go through *. The only problem though is for ATA 186. They need canreinvite = No when they are in a NAT environment. - Original Message - From: "Low, Adam" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Monday, July 28, 2003 11:29 AM Subject: [Asterisk-Users] RTP session traversing Asterisk server ... > > I've been reading up on the SIP and related (SDP/RTP) RFC's and as I would expect the RTP session should ideally be between the two end points of the call, in my case the AS5300 and the 7940 which are connected on the same VLAN as the Asterisk server. > > When I sniff the packets on the VLAN I find that all RTP packets are being relayed by the Asterisk server causing increased load on the server and ultimately a higher latency between the two end points. > > Is this a typical operation of Asterisk or is this possibly due to the fact that some of the phones (not those used in the tests) are running NAT and Asterisk relays all RTP packets ? > > Adam > > > * DISCLAIMER * > > This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] g729 Codec
Ricardo Have you tested g729 between two endpoints (SIP) for over 5mins? My experience has been that after 3-4 mins both ends begin to get huge delays and after a few minutes is impossible to continue the conversation. HAve you done any testing similar to mine? - Original Message - From: "Ricardo Villa" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]>; <[EMAIL PROTECTED]> Sent: Monday, July 28, 2003 5:10 PM Subject: Re: [Asterisk-Users] g729 Codec > Thanks Wipeout. I ordered a couple of licenses and have them running in the > lab. The codec works pretty good so far. > > I noticed that the transmitt packet time of the g.729 codec seems to be > hardcoded at 20ms. Is there anyway to adjust that via a config file? Most > implementations allow you to adjust it between 10-60ms. > > Thanks, > Ricardo Villa > http://www.telesip.net > > - Original Message - > From: "WipeOut ." <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > Sent: Monday, July 28, 2003 2:04 AM > Subject: Re: [Asterisk-Users] g729 Codec > > > > Its just like any other codec so it should work in SIP, IAX or any other > connection.. > > > > > Hi, > > > > > > Do the g729 codec licenses for Asterisk work on a SIP environment (only > SIP UAs running g729 + Asterisk)? I would like to buy a couple for a SIP > test lab but I have not found any documentation on wether it works for SIP > UAs or not. The Digium page only mentions: "The G.729 codec works with all > Digium cards." > > > > > > Can somebody tell me please? > > > > > > Thanks, > > > Ricardo Villa > > http://www.telesip.net > > -- > > __ > > http://www.linuxmail.org/ > > Now with e-mail forwarding for only US$5.95/yr > > > > Powered by Outblaze > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] executing an agi script after a successful Dial
Thanks for the response. In addition to what you stated, I think there is
another problem with Asterisk::AGI
This is the test script
#!/usr/bin/perl
use Asterisk::AGI;
$AGI = new Asterisk::AGI;
my %input = $AGI->ReadParse();
my $num = $AGI->get_variable('FOO')
$AGI->verbose("get_variable\"FOO\"=$num",1);
--
extensions.conf
exten=> h, 1,SetVar(FOO=)
exten=> h,2,Agi,test.agi
exten => _6XX,1,Agi,db.agi
exten => _4XX,1,Dial,${TEST}
--
If I call the Agi by dialing 666 the perl script works just fine and it runs
twice (I think this is strange since I didn´t execute a Dial)
If I dial 444 the script executes but I get no output.
Therefore it seems there is a problem with Asterisk::AGI
- Original Message -
From: "Klaus-Peter Junghanns" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Friday, July 25, 2003 8:32 PM
Subject: Re: [Asterisk-Users] executing an agi script after a successful
Dial
Hi Dan,
no wonder. when the h extension is called the channel (including all
the channel variables you want to read with get_var) is gone. pass the
channel variables you need to acces as an argument to the agi script,
e.g.: exten => h,1,AGI(myagi.agi,${EXTEN} ${CALLERIDNUM})
regards
kapejod
--
Klaus-Peter Junghanns
CEO,CTO
Junghanns.NET GmbH
Breite Strasse 13 - 12167 Berlin - Germany
fon: +49 30 79705392
fax: +49 30 79705391
iaxtel: 1-700-157-8753
email: [EMAIL PROTECTED]
http://www.junghanns.net/asterisk
Am Sam, 2003-07-26 um 01.28 schrieb Dan Fernandez:
> John
>
> Thanks for the response. This seems to be what I am looking. However, I
> have discovered a problem with a simple perl script triggered from the h
> extension.
>
> I am using perl-Asterisk and if I call the script from any extension in
> works fine. However, if I call the same script from h the get_variable and
> verbose functions don´t work anymore.
>
> Rgds
> Dan
> - Original Message -
> From: "John Todd" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent: Wednesday, July 23, 2003 8:20 PM
> Subject: Re: [Asterisk-Users] executing an agi script after a successful
> Dial
>
>
> > >On Wednesday 23 July 2003 05:04 pm, Dan Fernandez wrote:
> > >> I would like to run an agi script (to calculate the cost of a long
> > >> distance or international call) right after I execute a Dial app.
> > >> Can this be configured in extensions.conf? It seems the entries
> > >
> > >It cannot. If the Dial app succeeds in getting a connected channel,
> > >it will ALWAYS return -1, which signals a hangup to Asterisk. The
> > >only time Dial will ever return control to the dialplan is if either
> > >the channel is not available or if the channel does not get connected.
> >
> > Hmm... I'm not so sure about what the question was, and if perhaps
> > there is some confusion about what is desired here. In my example
> > configs, I use the "h" extension to clean up call recording after
> > Dial has terminated. Seems to work for me, but perhaps it's not
> > supposed to work. :)
> >
> > Dan - try putting your routines in an extension called "h". This may
> > get executed after Dial terminates normally or abnormally.
> >
> > JT
> >
> >
> > > > right after a Dial app get executed only if the Dial app was
> > >> executed unsucessfully. Would I have to execute the dial app from
> > >> the agi script?
> > >
> > >No, again, the Dial app won't return control to the AGI script until
> > >after the call is complete. You're pretty much going to have to do
> > >whatever you want to do prior to executing Dial or after the call is
> > >complete. Of course, you could create a separate thread which
> > >runs parallel to the channel thread and does various monitoring
> > >tasks, but that would require some C programming skills.
> > >
> > >-Tilghman
> > >
> > >___
> > >Asterisk-Users mailing list
> > >[EMAIL PROTECTED]
> > >http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> > ___
> > Asterisk-Users mailing list
> > [EMAIL PROTECTED]
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
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Re: [Asterisk-Users] executing an agi script after a successful Dial
John Thanks for the response. This seems to be what I am looking. However, I have discovered a problem with a simple perl script triggered from the h extension. I am using perl-Asterisk and if I call the script from any extension in works fine. However, if I call the same script from h the get_variable and verbose functions don´t work anymore. Rgds Dan - Original Message - From: "John Todd" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Wednesday, July 23, 2003 8:20 PM Subject: Re: [Asterisk-Users] executing an agi script after a successful Dial > >On Wednesday 23 July 2003 05:04 pm, Dan Fernandez wrote: > >> I would like to run an agi script (to calculate the cost of a long > >> distance or international call) right after I execute a Dial app. > >> Can this be configured in extensions.conf? It seems the entries > > > >It cannot. If the Dial app succeeds in getting a connected channel, > >it will ALWAYS return -1, which signals a hangup to Asterisk. The > >only time Dial will ever return control to the dialplan is if either > >the channel is not available or if the channel does not get connected. > > Hmm... I'm not so sure about what the question was, and if perhaps > there is some confusion about what is desired here. In my example > configs, I use the "h" extension to clean up call recording after > Dial has terminated. Seems to work for me, but perhaps it's not > supposed to work. :) > > Dan - try putting your routines in an extension called "h". This may > get executed after Dial terminates normally or abnormally. > > JT > > > > > right after a Dial app get executed only if the Dial app was > >> executed unsucessfully. Would I have to execute the dial app from > >> the agi script? > > > >No, again, the Dial app won't return control to the AGI script until > >after the call is complete. You're pretty much going to have to do > >whatever you want to do prior to executing Dial or after the call is > >complete. Of course, you could create a separate thread which > >runs parallel to the channel thread and does various monitoring > >tasks, but that would require some C programming skills. > > > >-Tilghman > > > >___ > >Asterisk-Users mailing list > >[EMAIL PROTECTED] > >http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] executing an agi script after a successful Dial
I just want to run a script to calculate the cost of a call to a cell phone,long distance, etc. right after I execute a Dial app. and the call is complete. I gathered from your response that it would be possible to execute the dial from inside the agi but this is probably not the way to go. Aside from coding in c, or running a script with cron, what other alternatives are there? Someone must have done something on this regard (calculating call costs) - Original Message - From: "Tilghman Lesher" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Wednesday, July 23, 2003 7:18 PM Subject: Re: [Asterisk-Users] executing an agi script after a successful Dial > On Wednesday 23 July 2003 05:04 pm, Dan Fernandez wrote: > > I would like to run an agi script (to calculate the cost of a long > > distance or international call) right after I execute a Dial app. > > Can this be configured in extensions.conf? It seems the entries > > It cannot. If the Dial app succeeds in getting a connected channel, > it will ALWAYS return -1, which signals a hangup to Asterisk. The > only time Dial will ever return control to the dialplan is if either > the channel is not available or if the channel does not get connected. > > > right after a Dial app get executed only if the Dial app was > > executed unsucessfully. Would I have to execute the dial app from > > the agi script? > > No, again, the Dial app won't return control to the AGI script until > after the call is complete. You're pretty much going to have to do > whatever you want to do prior to executing Dial or after the call is > complete. Of course, you could create a separate thread which > runs parallel to the channel thread and does various monitoring > tasks, but that would require some C programming skills. > > -Tilghman > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] executing an agi script after a successful Dial
I would like to run an agi script (to calculate the cost of a long distance or international call) right after I execute a Dial app. Can this be configured in extensions.conf? It seems the entries right after a Dial app get executed only if the Dial app was executed unsucessfully. Would I have to execute the dial app from the agi script? Thanks in advance. Dan
Re: [Asterisk-Users] Problems with g729
Martin I just updated the new codec_g729.so. The problem with the delay between to SIP endpoints it is still there. That is, after 3-4 mins, the delay begins to get really bad. With g723 under the same conditions I have no problem. Any idea what the problem could be? Rgds Dan - Original Message - From: "Martin Pycko" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Wednesday, July 23, 2003 4:47 PM Subject: Re: [Asterisk-Users] Problems with g729 Try the new_codec_binary/codec_g729b.so from the digium ftp site. regards Martin On Wed, 23 Jul 2003, Dan Fernandez wrote: > I am having some problems with g729 with SIP and ZAP channels. > > 1) > I have two g729 licences. Very frequetnly (I don´t know what triggers the error) I get the following warnings and error when I try to place a call via SIP to my X100P. The only way to get out of this is through a restart of *. When the error ocurrs there are no other calls in place. Any ideas? > > > Error Opening channel:2 not available, see va_g729_init_global(..) WARNING[71694]:File codec_g729b.c line 102 (g729lin_new): No available g729b resource for channel 2 > WARNING:[71694] File translate.c Line 111 (ast_translator_build_path):Failed to build translator path from 8 to 6 Zap1-1 answered SIP/105-ce3c > WARNING[71694]: File chan_zap.c Line 3367 (zt_write):Cannot handle frames in 256 format > Hangup Zap/1-1 > > > 2) > have discovered a problem when using g729 under the following setup: > > SIP call between a Budgetone 102 and ATA 186 (configured without silence suppresion). Both ends have a ADSL 64kbps. Both ends are behind Linksys routers. The pings between them are aprox. 100ms. No other local users on each end. * is being hosted on a PIII,128MB. No other calls are being handled at the time of the test. > > Basically, after a few minutes, with g729, both ends consistently start getting delays up to a point where it becomes almost unbearable to speak. If we switch to g723 the problem goes away. > > ANy ideas what´s going on? > > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problems with g729
I am having some problems with g729 with SIP and ZAP channels. 1) I have two g729 licences. Very frequetnly (I don´t know what triggers the error) I get the following warnings and error when I try to place a call via SIP to my X100P. The only way to get out of this is through a restart of *. When the error ocurrs there are no other calls in place. Any ideas? Error Opening channel:2 not available, see va_g729_init_global(..) WARNING[71694]:File codec_g729b.c line 102 (g729lin_new): No available g729b resource for channel 2 WARNING:[71694] File translate.c Line 111 (ast_translator_build_path):Failed to build translator path from 8 to 6 Zap1-1 answered SIP/105-ce3c WARNING[71694]: File chan_zap.c Line 3367 (zt_write):Cannot handle frames in 256 format Hangup Zap/1-1 2) have discovered a problem when using g729 under the following setup: SIP call between a Budgetone 102 and ATA 186 (configured without silence suppresion). Both ends have a ADSL 64kbps. Both ends are behind Linksys routers. The pings between them are aprox. 100ms. No other local users on each end. * is being hosted on a PIII,128MB. No other calls are being handled at the time of the test. Basically, after a few minutes, with g729, both ends consistently start getting delays up to a point where it becomes almost unbearable to speak. If we switch to g723 the problem goes away. ANy ideas what´s going on?
[Asterisk-Users] Delays with g729 and SIP. How come?
I have discovered a problem when using g729 under the following setup: SIP call between a Budgetone 102 and ATA 186 (configured without silence suppresion). Both ends have a ADSL 64kbps. Both ends are behind Linksys routers. The pings between them are aprox. 100ms. No other local users on each end. * is being hosted on a PIII,128MB. No other calls are being handled at the time of the test. Basically, after a few minutes, with g729, both ends consistently start getting delays up to a point where it becomes almost unbearable to speak. If we switch to g723 the problem goes away. ANy ideas what´s going on? Dan
Re: [Asterisk-Users] Billsec on CDR
Steve Can you please give us some guidance on how to make call progress work outside the US or UK? Thanks Dan - Original Message - From: "Stephen Davies" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Saturday, June 21, 2003 4:51 AM Subject: Re: [Asterisk-Users] Billsec on CDR > > > On Fri, 20 Jun 2003, Tan Aks wrote: > > > Isn't there any way to make callprogress work for people in Europe? What is > > it that is needed to make it work? > > I've done call progress for the UK. Patch to the -dev list by the end > of the weekend. > > What country do you want? > > Steve > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Budgetone and Voicemail
Yes! It did work with g729 and dtmfmode=info. Thanks a lot! - Original Message - From: "Michael Bielicki" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Wednesday, July 09, 2003 7:35 PM Subject: Re: [Asterisk-Users] Budgetone and Voicemail > try dtmfmode=info > solved all my former problems and even works with g723.1 > :) > On Wednesday 09 Jul 2003 11:15 pm, Dan Fernandez wrote: > > I have a Budgetone 102 with the latest firmware 1.0.3.72 and using > > dtmfmode=rfc2833 > > > > With g711 I have no problem with Voicemail or Voicemail2. > > > > With g729 it always repeats digits and it is impossible to check my > > voicemail (or any other apps that require digits) > > > > > > - Original Message - > > From: "WipeOut ." <[EMAIL PROTECTED]> > > To: <[EMAIL PROTECTED]> > > Sent: Tuesday, July 08, 2003 3:42 PM > > Subject: Re: [Asterisk-Users] Budgetone and Voicemail > > > > > I have had double digits being passed every now and then once I am into > > > > voicemail.. I haven't had a problem with the initial login stage.. I also > > haven't had time to look into it yet.. > > > > > You could try changing the DTMF mode and see if it helps.. > > > > > > Later.. > > > > > > > I have a problem with using voicemail on the Budgetone phones. When > > > > entering the mailbox and password, sometimes some keys will register > > > > multiple times (as shown on console when it says no such user in config > > > > file) and sometimes some keys won't even register at all. It seems > > > > totally random. Has anyone seen this problem? Any recommendations > > > > would be greatly appreciated. Thanks. > > > > > > > > > > > > Brian Borders > > > > [EMAIL PROTECTED] > > > > > > > > ___ > > > > Asterisk-Users mailing list > > > > [EMAIL PROTECTED] > > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > -- > > > __ > > > http://www.linuxmail.org/ > > > Now with e-mail forwarding for only US$5.95/yr > > > > > > Powered by Outblaze > > > ___ > > > Asterisk-Users mailing list > > > [EMAIL PROTECTED] > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Budgetone and Voicemail
I have a Budgetone 102 with the latest firmware 1.0.3.72 and using dtmfmode=rfc2833 With g711 I have no problem with Voicemail or Voicemail2. With g729 it always repeats digits and it is impossible to check my voicemail (or any other apps that require digits) - Original Message - From: "WipeOut ." <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Tuesday, July 08, 2003 3:42 PM Subject: Re: [Asterisk-Users] Budgetone and Voicemail > I have had double digits being passed every now and then once I am into voicemail.. I haven't had a problem with the initial login stage.. I also haven't had time to look into it yet.. > > You could try changing the DTMF mode and see if it helps.. > > Later.. > > > I have a problem with using voicemail on the Budgetone phones. When > > entering the mailbox and password, sometimes some keys will register > > multiple times (as shown on console when it says no such user in config > > file) and sometimes some keys won't even register at all. It seems > > totally random. Has anyone seen this problem? Any recommendations > > would be greatly appreciated. Thanks. > > > > > > Brian Borders > > [EMAIL PROTECTED] > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > -- > __ > http://www.linuxmail.org/ > Now with e-mail forwarding for only US$5.95/yr > > Powered by Outblaze > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco ATA-186 config guide for Asterisk
John, Thanks for the detailed guide. As you mentioned, the situation where two ATAs behind NAT want to establish a direct connection is not resolved yet. In that case, the canreinvite would have to be set to no and some other solution outside of * would have to be used to traverse the NAT. Have you tested any alternatives? Rgds Dan - Original Message - From: "John Todd" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Sunday, June 29, 2003 7:35 PM Subject: [Asterisk-Users] Cisco ATA-186 config guide for Asterisk > > I really should be doing something better on this beautiful weekend, > but I'm trying to save myself some time for later projects by > documenting some things that have been particularly troublesome in > the past. That being said... > > I've written up a configuration guide for the Cisco ATA-186, which > describes some of the features that are possible to set in the ATA > and specifically what needs to be done to get it working with > Asterisk. > > It's not pretty, it's not HTML, but it's a lot of hints that I've > collected from the list and other sources over the last year or so: > > http://www.loligo.com/asterisk/Cisco/ATA-186-guide.v20030628.txt > > > JT > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Transcoding
I have a Budgetone and an ATA but none of them support GSM. I´d like to place call to the PSTN with my X100P via a WAN (64kbps). g711 is out of the question. Can * transcode from g723.1 to GSM? How costly is it? I have tried different configurations on sip.conf and extensions.conf but have had no luck. Is this transcoding possible?
[Asterisk-Users] Billsec on CDR
I have an X100P and when I place calls to the PSTN which are not answered, the Billsec field of the CDR still logs the seconds that the phone rang. Can someone please confirm that this has to do with the ringcadance of the indications.conf file? Is there anything else I need to check ? Thanks in advance
Re: [Asterisk-Users] Budgettone 100 phone Configuration
Will look into this once someone can help me with the configuration behind NAT (without NAT I have no problem) I am using v1.0.3.53 and a linksys router (the phone IP is 192.168.1.2) I´ve tried in my sip.conf with and without NAT=1. In the phone, if I set the outbound proxy to the linksys it doesn´t do anything. If I leave outbound proxy empty it registers and I can place calls but no audio either way. I have also tried setting the phone for NAT and no NAT (no STUN server). Don´t know what else to try. Can someone please help me? - Original Message - From: "Greg Renouf" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Thursday, June 05, 2003 8:16 PM Subject: RE: [Asterisk-Users] Budgettone 100 phone Configuration > I'm using v.1.0.3.58 and am experiencing that my phone crashes every > time the call reaches about 45 minutes in length. > > Has anybody had a similar experience? > > -GSR > > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Stephen R. > Besch > Sent: 05 June 2003 19:03 > To: [EMAIL PROTECTED] > Subject: re: [Asterisk-Users] Budgettone 100 phone Configuration > > The updated Budgetone firmware (1.0.3.60) has indeed fixed the "silent > DTMF" issue. > > >By the way, Grandstream just got the "silent DTMF" problem fixed for > me > >and sent me an updated revision this morning (1.0.3.60). I am just > >about to install it, but it may require that I debug my tftp server, > >which I haven't tested yet. I'll post the list as soon as I get the > >new version loaded. > -- > Stephen R. Besch, Ph.D. > SachsLab > 320 Cary Hall > SUNY at Buffalo > Buffalo, NY 14214 > (716) 829-3289 x106 > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] howto reduce number of rings ?
Is there a way to reduce the number of rings if there is a message on the mailbox. That is I set the Wait app to 10 secs but then want it to pick up a call right away after someone leaves a message (ie I am not at home, office) How can i do this? Thanks in advance Dan
Re: [Asterisk-Users] iconnect quality?
I found similar problems. With my phonejack I can make a call with ulaw with decent quality (I have a 64k line). However, with Messenger I hear a brief horrible noise and that´s it. - Original Message - From: "Jim Archer" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Tuesday, March 11, 2003 8:17 PM Subject: Re: [Asterisk-Users] iconnect quality? > Ok! When I use the prefix and I allow gsm it does work! And the > quality is fine. > > There are two problems we're having now. > > 1 - From watching the udp fly by, it seems that iconnect does not know when > we hang up. For example, if I call a voice mail and it starts giving me > its speal and I hang up, iconnect stays connected until the VM hangs up at > its end. > > Next, if we try to call out via iconnect from a sip client extension (like > a windows soft phone) all we hear is horrible noise. > > Has anyone else had these issues? > > Jim > > > --On Tuesday, March 11, 2003 3:34 PM -0500 Gregg Lebovitz > <[EMAIL PROTECTED]> wrote: > > > I haven't play around enough to know whether or not the prefix is a > > toggle. I will do some experimenting and let you know. Right now I am > > prefixing all my calls with . > > > > My experience is that when the carrier's format is G723.1, you can't > > hear the incoming voice. When it is in G711 you can. I have made several > > calls using G711 and they are acceptable quality. Note that if you > > disallow=gsm in the sip.conf file you will get the 488 media errors you > > reported earlier. > > > > Below are my config files for sip and the linejack cards: > > > > ; > > ; SIP Configuration for Asterisk > > ; > > [general] > > port = 5060 ; Port to bind to > > bindaddr = 0.0.0.0 ; Address to bind to > > context=iconnect ; Default for incoming calls > > allow=gsm > > allow=ulaw > > allow=alaw > > > > ;register=1813342:[EMAIL PROTECTED] > > ;register=1202454:[EMAIL PROTECTED] > > > > [iconnecthere] > > type=friend > > username= > > secret=XXX > > host=sipauth.deltathree.com > > > > ; > > ; Linux Telephony Interface > > ; > > ; Configuration file > > ; > > [interfaces] > > > > mode=dialtone > > format=ulaw > > echocancel=medium > > silencesupression=no > > > > context=local > > context=default > > > > txgain=100% > > rxgain=100% > > device => /dev/phone0 > > > > > > > > On Tue, 2003-03-11 at 14:28, Jim Archer wrote: > >> Hi Greg and thanks very much... > >> > >> A few questions... > >> > >> First, regarding the prefix, it seemed that this acts as a toggle, > >> switching from the one codec to the other. But how do I set which me > >> system uses by default? Or does iconnect always use the high bandwidth > >> one by default (such that the always switches to the low bandwidth > >> one)? > >> > >> Next, I am still struggling to understand the SIP options and what goes > >> where. Could you please tell me where the format command goes? Is this > >> an option on the channel? I thing the allow goes in sip.conf. > >> > >> Finally, does this have any impact on the problem where the person > >> called can not be heard? > >> > >> Thanks!!! > >> > >> Jim > >> > >> --On Tuesday, March 11, 2003 1:35 PM -0500 Gregg Lebovitz > >> <[EMAIL PROTECTED]> wrote: > >> > >> > Jim, > >> > > >> > I changed my extensions entry for iconnect to: > >> > > >> > exten => _1XX,1,Dial,SIP/[EMAIL PROTECTED] > >> > > >> > and my calls work fine with ulaw. I am calling from a linejack card > >> > with format=ulaw and SIP with allow=ulaw. > >> > > >> > Gregg > >> > > >> > On Mon, 2003-03-10 at 23:01, Jim Archer wrote: > >> >> --On Monday, March 10, 2003 4:47 PM -0300 Dan Fernandez > >> >> <[EMAIL PROTECTED]> wrote: > >> >> > >> >> > Iconnect uses codecs g723 and g711 that can be configured for each > >> >> > account (you can change them by the prefix) > >> >> > >> >> I tried adding the in front of a number and it reliably generates > >> >> error "488 invalid media." > >> >> > >> >> > >> >> ___
Re: [Asterisk-Users] segfault WAS astman make problems
I was able to install the rpms but when I run astman I get a segfault after I try to login (independently of the user I use) Yesterday I saw another posting regarding a segfault with astman. Any suggestions? - Original Message - From: "Michiel Betel" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Tuesday, March 11, 2003 2:22 PM Subject: RE: [Asterisk-Users] astman make problems > Newt is no longer included in SuSE 8.1, I tried installing the 7.2 newt > packages but they don't work correctly, finally I installed the 7.2 source > rpms and rebuilt them for 8.1, that works > > Michiel > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Ben Klang > Sent: dinsdag 11 maart 2003 8:52 > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] astman make problems > > > The newt development package can be installed via yast2. > > run 'kdesu yast2' or 'su -c yast2', find the Install Packages option, and > search for newt. > > Hope that helps, > > -BAK > > On Mon, 2003-03-10 at 16:26, Dan Fernandez wrote: > > > > Can astman be compiled without newt? I have Suse 8.1 and it doesn´t > > have newt. If needed, where can I get it? > > > > Thanks in advance > -- > Ben Klang <[EMAIL PROTECTED]> > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iconnect quality?
Yes, 64K is not much of a broadband and believe it or not, I am paying US$60 for it (I believe this is the case in many 3rd world countries). Is there a codec translator between GSM and g723? How come I can use FWD just fine with g711? - Original Message - From: "Steven Critchfield" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Monday, March 10, 2003 5:33 PM Subject: Re: [Asterisk-Users] iconnect quality? > On Mon, 2003-03-10 at 13:47, Dan Fernandez wrote: > > Iconnect uses codecs g723 and g711 that can be configured for each account > > (you can change them by the prefix) > > > > With their dialer and g723 I can here just fine (I have a 64k broadband > > connection). With their dialer and g711 the quality suffers greatly. > > > > With * and GSM I cannot here anything (and don´t know if they can here me). > > The call gets logged on iconnect´s CDR. Upon looking at the SIP debug > > everything appears just fine, but again, I cannot hear anything. > > g711 is 8 bit 8khz, or 64Kbit of audio data alone without the overhead > of TCP/IP nor SIP. 64Kbit is not broadband, it is a DS0. It may be a tad > over dial up, but please don't consider it broadband. > > This would explain your quality problem, you fill the link in the first > set of samples, and the rest is queued up and therefore does not arrive > like a stream should. Each fram probably adds a millisecond or more to > the stream and soon enough you are so starved for audio data that it > should give up. > -- > Steven Critchfield <[EMAIL PROTECTED]> > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] astman make problems
Can astman be compiled without newt? I have Suse 8.1 and it doesn´t have newt. If needed, where can I get it? Thanks in advance
Re: [Asterisk-Users] iconnect quality?
Iconnect uses codecs g723 and g711 that can be configured for each account (you can change them by the prefix) With their dialer and g723 I can here just fine (I have a 64k broadband connection). With their dialer and g711 the quality suffers greatly. With * and GSM I cannot here anything (and don´t know if they can here me). The call gets logged on iconnect´s CDR. Upon looking at the SIP debug everything appears just fine, but again, I cannot hear anything. With * and ulaw or mlaw I can hear the first 5 secs and nothing else. I have tried FWD (not going through *) which uses g711 and I have had nor problem whatsoever. - Original Message - From: "Jim Archer" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Monday, March 10, 2003 8:51 AM Subject: [Asterisk-Users] iconnect quality? > Does anyone here use iconnect regularly with Asterisk? If so, what do you > think of its reliability and quality? I used up my 10 free minutes just > getting it to work. > > By the end it was working, but I found that (1) many calls did not connect, > due to a variety of errors reported by them (service unavailable and such) > and (2) when I did connect, I could be heard but I could not hear the > person I was talking to. > > I would appreciate hearing any experiences. > > Thanks! > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iconnecthere 480 error: is there a workaround?
I have tried with iconnect sofphone calling, hanging up, and calling immediately and did not have any problems as you do with *. I think the problem is with the * and not with iconnect. I get the 480 error more often than you do, and when I am able to connect, I can only hear the other ends voice for a few seconds with a horrible quality (I have tried with all the codecs) and get the following errors: Mar 6 05:49:20 DEBUG[5126]: File chan_sip.c, Line 3155 (handle_request): That's odd... Got a response on a call we dont know about. Mar 6 05:49:21 DEBUG[5126]: File chan_sip.c, Line 3155 (handle_request): That's odd... Got a response on a call we dont know about. Mar 6 05:49:44 DEBUG[5126]: File chan_sip.c, Line 3159 (handle_request): Ignoring out of order response 105 (expecting 101) Mar 6 05:49:48 DEBUG[5126]: File chan_sip.c, Line 3159 (handle_request): Ignoring out of order response 106 (expecting 101) debug and if I look at the sip debug I see a lot of INVITE restransmits. The bottom line is that I have not been able to make calls with iconnect yet. What codec are you using? When I first started using iconnect with their dialer (I have a 64k broadband connection) the call quality was very poor. I then contacted them and asked them to changed the codec and they did to g723. The call quality with their dialer is excelent! - Original Message - From: "John Todd" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Tuesday, March 04, 2003 1:09 AM Subject: Re: [Asterisk-Users] iconnecthere 480 error: is there a workaround? > I get these errors (480 "Temporarily...") when I try to use my > iconnect account quickly after hanging up on a previous session. > They have some sort of contention locking system which allows only > one call at a time on an account, and if you do not give it adequate > time to "settle", you'll hit that error type. I have found that > waiting 15 seconds or so before making another call will ensure > completion. > > Personally, I think they should let multiple sessions through on the > same account and have a "hard limit" set on consumablel minutes in a > monthly billing period. In other words, if their concern is about > fraud, then fine - make it such that the account holder must > "recharge" their account past a certain limit. Don't limit my > burning of minutes due to poorly contrived fraud protection schemes; > heck, you'd think they'd want customer to burn up minutes as quickly > as possible. > > JT > > > > >I am going to have to find a fix for this problem or I'm going to > >have to quit using iconnect. > > > >About one call in 10 or so, iconnect's gateway gives me an error > >(console output appended below). > > > >So upon receiving the error, which as a 4XX error means, "Fatal," > >asterisk gives up and drops the call. But not iconnect!! The phone > >at the other end starts ringing, and rings several times before the > >call is dropped. > > > >So the person at the other end, unless it's my friends who are now > >inured to this, wonder WTF is going on. > > > >I sent a mail to iconnect asking if they don't agree that it's > >broken, but in the near-term I need to find a fix. > > > >Thx. > > > >B. > > > >* > >Console output begins here, numbers elided to protect the innocent :-) > > > > -- Called [EMAIL PROTECTED] > > -- Got SIP response 480 "Temporarily not available" back from > >213.137.73.140 > > == No one is available to answer at this time > >WARNING[311310]: File pbx.c, Line 1179 (ast_pbx_run): Channel > >'SIP/ata1-2da9' sent into invalid extension '1XXXNNN' in > >context 'iconn', but no invalid handler > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
