Re: [Asterisk-Users] Grandstream 496 not working on cordless phone
On Mon, 2005-09-26 at 17:24 -0400, Nana Tandoh wrote: > Hi All, > > We are using SER/Asterisk, it works fine from X-lite to corded phones > but have problems using a cordless phone on the Handytone 496. Has > anyone experienced this problem. Well, if you told us what the problems are perhaps we could help. -- Dave Cotton <[EMAIL PROTECTED]> ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IBM x306 - some progress
Hi Marco, As far as I can recall, the IBM setup utility can enable you to change the IRQ of the SCSI controller. In addition, I've never seen a WildCard board bound to IRQ7 on any box, which is very weird in it self. I'm flying over to Ireland today (actually, at the airport right now), and I'm coming back on Sunday. If You'd like, you can bring your box to my office after Rosh-Hashana, and I'll try to help you out. Nir S -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Marco Supino Sent: Monday, September 26, 2005 10:06 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] IBM x306 - some progress Hi, I asked yesterday about a problem with x306 and IRQ sharing, didnt get much info, now, i was playing with lspci, and see something strange, lspci -v shows me the TDM400P card is on IRQ 7, and the SCSI card is also on IRQ 7, lspci -bv (from the man - b - shows "bus-centric view, as seen by the BUS and not by the kernel) shows me the TDM400P is on IRQ 5, why does the kernel puts it on IRQ 7 ? any insights much appriciated. Marco. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] "Non-blocking" Dial (and other commands): is there a way?
In order to use a with GrandStream BT-488 as "pass-through" gateway, I need a way of sending the FXO port off hook when I'm using the FXS port for VoIP communications, because I want to use the "hunting line" feature to let incoming call skip that FXO port and move on to the next free line. The only way I have found to engage a device without getting blocked until the call ends passes through an AGI script that drops a callfile into the /var/spool/asterisk/outgoing directory, telling Asterisk to dial the FXO port and then connect the channel to, say, the MusicOnHold() application. When I'm done, I can then issue a SoftHanghup() to the FXO device. This method strikes me as pretty clumsy: aren't there better ways of issuing commands from the dialplan in "detached mode", perhaps getting a handle useful to regain control later, and proceed to do other things? Enzo ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Bad FCS nightmare to Nortel SL100 with TE410P
I have an * box connected to a Nortel SL100 through a PRI (US) using the Digium TE410P (quad-span T1 card). I don't have access to the SL100 - it is handled by another group. The span comes up OK (timing, framing fine). However, as soon as the D channel comes up, I get endless "HDLC Bad FCS" errors. I modified logger.conf to get rid of the messages (so I could see what else was going on), and noticed that the B-channel restart was going horribly slow, and the D channel was essentially "flapping" up and down. I could sometimes squeeze a call in while the D channel was up, but it would only last a few seconds. I also get "short write" errors as well (unfortunately I don't have a log of these and can't get at the PRI at the moment to get the exact error message). I've had the physical circuit tested and there are no issues with it. In fact, it was working fine to the same switch as an E&M digital trunk up until we tried to change it to a PRI. I've tried 3 different TE410Ps on three different * versions (based on things I've seen in previous posts). All behave exactly the same. The versions are 1.0.5, 1.0.9, and a CVS version of 1.2.0-beta1 pulled down at the end of August. In all cases, the systems are Dell PowerEdge 1750s (using RAID, no IDE drives involved) on Debian / kernel 2.4.27. I see no indication of problematic interrupts. In one test, there were 3 other PRIs running on the TE410P (in production) and there are no problems with any other PRIs. Ditto the configuration (I've checked and am doing the exact same thing with all my PRIs, just on different channels). Before I start providing configuration excerpts - has anyone had this problem connecting to an older Nortel Meridian switch and if so, what did you do to fix it? I suspect that there is a subtle configuration option on the SL100 that is wrong, but since I don't have access to it I can't confirm that. Can the wrong switch type cause FCS errors? Is there anything specific I can look at? For those who speak SL100, do you know of any specific parameter I can point the SL100 guy to? One more data point: I threw the PRI from the SL100 onto a spare port on a Cisco AS5350 and the AS5350 isn't complaining (no frame slips, no problem with the D channel). I'm pulling my hair out with this. Any help or pointers to info would be helpful. I will post a summary to the list if I get any useful private e-mail about this. Thanks! --- Gil Kloepfer [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SPA-3000 and incoming faxes
If you are willing to dedicate a fax line and forward faxes to a dedicated extension it work 100% for Incoming and Outgoing faxes with Sipura-3000 Though, you have to change in the Regional Tab: Ring Waveform: from Sinusoid to Trapezoid I've been using NVbackgroundDetect with Sipura-3000 and forwarding the to fax extension (Hylafax) if it is a fax and voice line if it is a voice call. So far I've been receiving faxes from Europe, Asia, USA and it work 98% most of the time. I have only one customer in Mexico and one in Asia that NVbackgroundDetect has a problem with to recognize fax signal. -- #Joseph On Mon, 2005-09-26 at 23:08 -0500, Tim Litwiller wrote: > I've been running with a generic X100P for 5 or so months and every once > in a while I have problem receiving faxes. I see that others have the > same problems and some worse than I have with these boards so I was > wondering if using a Sipura SPA-3000 would be any more reliable. > > Has anyone had enough experience to tell me if that would definitely fix > the random fax error. > > PS. I have * at home and have it configured to send faxes to an > extension that has a fax machine connected to it and the fax machine is > set to auto answer the first ring. It is connected to a SPA-2002 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] re: DTMF woes, continued
I'm afraid this may not be helpful, but I will try, When I was working at a previous company, they where just starting to switch everything over to VoIP, since I was the new guy, I got one of the VoIP lines because of not having any more "real" phone lines. I had the same problem, the Asterisk guy fixed it by changing the way DTMF tones were sent in the phone settings, from InBand to one of the others, and made a change in Asterisk. After this, I had no more problems. I hope this can help you with a starting point. Kevin -Original Message- From: Esteban Guana-Jarrin [mailto:[EMAIL PROTECTED] Sent: September 26, 2005 9:27 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] re: DTMF woes, continued Hi Yair, Please let me if you managed to fix the DTMF tone issue, which you were experiencing couple of months ago. If not can you share any advancement. I'm currently experiencing the same issue, I can make outbound calls but DTMF will not work when dialing IVRs. My configuration is [EMAIL PROTECTED] 1.5, registering to Voip provider (Symbio), codec is g.729 and dtmf mode is set to rfc2833. Your help will be much appreciated Thanks & Regards, Esteban _ View 1000s of pictures, profiles and more now at Lavalife http://lavalife.com.au ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] re: DTMF woes, continued
You may interested to know that a lot of connections will need inband when speaking server to server. My system runs all users on RFC2833, except for other asterisk servers. They run inband, because otherwise the DTMF wasn't working. Just my 0.02 ->-Original Message- ->From: [EMAIL PROTECTED] ->[mailto:[EMAIL PROTECTED] On Behalf Of ->Esteban Guana-Jarrin ->Sent: Monday, September 26, 2005 10:27 PM ->To: asterisk-users@lists.digium.com ->Subject: [Asterisk-Users] re: DTMF woes, continued -> ->Hi Yair, -> ->Please let me if you managed to fix the DTMF tone issue, ->which you were experiencing couple of months ago. If not can ->you share any advancement. -> ->I'm currently experiencing the same issue, I can make ->outbound calls but DTMF will not work when dialing IVRs. My ->configuration is [EMAIL PROTECTED] 1.5, registering to Voip ->provider (Symbio), codec is g.729 and dtmf mode is set to rfc2833. -> ->Your help will be much appreciated -> -> ->Thanks & Regards, -> ->Esteban -> ->_ ->View 1000s of pictures, profiles and more now at Lavalife ->http://lavalife.com.au -> ->___ ->--Bandwidth and Colocation sponsored by Easynews.com -- -> ->Asterisk-Users mailing list ->Asterisk-Users@lists.digium.com ->http://lists.digium.com/mailman/listinfo/asterisk-users ->To UNSUBSCRIBE or update options visit: -> http://lists.digium.com/mailman/listinfo/asterisk-users -> ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: R: [Asterisk-Users] Problem setting up TDM22B card
On Mon, Sep 26, 2005 at 10:00:10PM -0400, Patrick wrote: > > I know it seems basic but did you make sure and plug power into the > board when you installed it into the PCI slot? I spent about three hours > trying to get the dang thing to work in my machine until I decided to > stick the card into another PCI slot. That is when I noticed that I had > forgotten to ALSO plug power into the board from the power supply. > Everything worked fine after that (yep, I was a noob). :-) You get a messge about it from the module at module load time. rmmod wctdm (or wcfxo) and re- modprobe it, and then run: dmesg |tail -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX provider w/Toronto & Detroit termination
Can anyone recommend a good IAX provider offering numbers in Toronto and Detroit? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Teliax
Does anyone have any experience with Teliax for inbound IAX? Jason ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SPA-3000 and incoming faxes
Tim We have used a SPA3000 with asterisk 1.0.9 and an E1, recieves fine, sending is impossible. Regards Mark -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tim Litwiller Sent: Tuesday, 27 September 2005 2:08 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] SPA-3000 and incoming faxes I've been running with a generic X100P for 5 or so months and every once in a while I have problem receiving faxes. I see that others have the same problems and some worse than I have with these boards so I was wondering if using a Sipura SPA-3000 would be any more reliable. Has anyone had enough experience to tell me if that would definitely fix the random fax error. PS. I have * at home and have it configured to send faxes to an extension that has a fax machine connected to it and the fax machine is set to auto answer the first ring. It is connected to a SPA-2002 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SPA-3000 and incoming faxes
I've been running with a generic X100P for 5 or so months and every once in a while I have problem receiving faxes. I see that others have the same problems and some worse than I have with these boards so I was wondering if using a Sipura SPA-3000 would be any more reliable. Has anyone had enough experience to tell me if that would definitely fix the random fax error. PS. I have * at home and have it configured to send faxes to an extension that has a fax machine connected to it and the fax machine is set to auto answer the first ring. It is connected to a SPA-2002 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] StripMSD or extension parser bug?
>From Show application stripMSD
StripMSD(count): Strips the leading 'count' digits from the channel's
associated extension. For example, the number 5551212 when stripped with
a
count of 3 would be changed to 1212. This app always returns 0, and the
PBX
will continue processing at the next priority for the *new* extension.
So, for example, if priority 3 of 5551212 is StripMSD 3, the next step
executed will be priority 4 of 1212. If you switch into an extension
which
has no first step, the PBX will treat it as though the user dialed an
invalid extension.
It is expecting the next step to be:
_XX,2,Dial(Zap/g1/${EXTEN})
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Jim Gottlieb
> Sent: Monday, September 26, 2005 11:28 PM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] StripMSD or extension parser bug?
>
> For years we've had the following simple context for outgoing calls:
>
> [outtrunk]
> ; match any NANP, and strip leading 1 off exten =>
> _1XX,1,StripMSD,1 ; dial outbound on trunk group 1
> exten => _XX,2,Dial,Zap/g1/${EXTEN}
>
>
> But when I upgraded on Friday to the latest CVSHEAD, this no
> longer works. If I send 13115552368 to this context, I get a
> message like
>
> pbx.c: Channel 'Zap/361-1' sent into invalid extension
> '3115552368' in context 'outtrunk', but no invalid handler
>
> I tried adding a separate line to match 10D:
>
> exten => _XX,1,Dial,Zap/g1/${EXTEN}
>
> but the same call generated a "timeout".
>
> I don't know if this is a bug in StripMSD, extension parsing,
> or user error.
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[Asterisk-Users] ICD with asterisk
Dear list, I have trying to work with the ICD with Asterisk Strange i could not able to even compile it . I have followed the ICD readme but no use ... ? And the ICD list in sourceforge seems to not much active.. ? Any suggestions are welcome with regards rk Yahoo! India Matrimony: Find your partner now.___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk fifo
Dear list, I want to use the asterisk fifo channel to be integrated with other applications in the asterisk like MoH . Do we have any implementations as such Basically what does we mainly use this asterisk fifo ? your views will be highly appreciated. with regards rk Yahoo! India Matrimony: Find your partner online.___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] StripMSD or extension parser bug?
For years we've had the following simple context for outgoing calls:
[outtrunk]
; match any NANP, and strip leading 1 off
exten => _1XX,1,StripMSD,1
; dial outbound on trunk group 1
exten => _XX,2,Dial,Zap/g1/${EXTEN}
But when I upgraded on Friday to the latest CVSHEAD, this no longer
works. If I send 13115552368 to this context, I get a message like
pbx.c: Channel 'Zap/361-1' sent into invalid extension '3115552368' in
context 'outtrunk', but no invalid handler
I tried adding a separate line to match 10D:
exten => _XX,1,Dial,Zap/g1/${EXTEN}
but the same call generated a "timeout".
I don't know if this is a bug in StripMSD, extension parsing, or user
error.
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[Asterisk-Users] Flash Panal
Ladies and Gentleman, Could someone point me in the direction AMP add-ons? I havve the ones that were referanced in the AMP instal PDF, but I cannot find the | Maintenance | addon like is found in [EMAIL PROTECTED] Does the Panal in AMP tell you anthing or do you have to some how login to it before it displays any info..What info can I expect. Thank you for your time Tommy ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] BRI Hunting, using both channels on one msn
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Saturday, September 24, 2005 2:30 PM To: Gregory Wiktor - ADCom Corp. Cc: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] BRI Hunting, using both channels on one msn On Fri, 23 Sep 2005 [EMAIL PROTECTED] wrote: > Hello Armin, > I tried your new version of chan capi and it works well. > > I did have one question about capi.conf. I have a bri with 2 spids, > but I want to have the second go to a zap fax channel. > > Right now I can direct it, but the echo canceller is setting up. Do > you know a way to cancel it? Fax works, but I suspect it would work > better with EC off on large faxes. I don't know enough about the spid stuff, but you should be able to create two interfaces in capi.conf (instead of one), which devices=1 for each. So you should be able to set different settings for each channel. Ok, I managed to setup two but had devices=1 When using fax via capi with Eicon cards, the echo canceler should not be used automatically. This is correct, however when I do it on an analog machine connected to a zap channel is what I am concerned about. I managed to also get a local modem to work via the zaptel port, and connect at 50kps, although asterisk kept mentioning a fax detect while the modem would connect. > I tried the capi fax receive, but the images came out with the wrong > dimensions(on .05). Maybe a problem with setting fine/normal resolution? Turned out to be photoshop, another viewer worked ok. > Also, > Is there a way to split each msn into a different call group in > capi.conf? I tried a few combinations but no luck. > I was thinking I could disable the EC for the line in general. See above. > Oh as per the hunt, I had verizon program a hunt into the line and it > seems to work now. It is funny though, since I think my usrobotics > modem can also do it, I just don't know exactly how it is handled. > capi just causes the line to report a busy. > > And there is a new eicon driver that works on 2.6. Which one do you mean? The new driver? Came out early sept. Now the scripts work on debian and you don't need to manually initialize it. Armin > Thanks, > Greg > -Original Message- > From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] > Sent: Tuesday, August 16, 2005 4:49 AM > To: Gregory Wiktor - ADCom Corp. > Subject: RE: [Asterisk-Users] BRI Hunting, using both channels on one > msn > > Hi Greg, > > now I understand. You use NI-1 with spids. I'm sorry, I don't know > anything about this protocol. ETSI does not have this 'channel-problem'. > > Maybe it can be solved with some load parameters for the BRI card. > You should use the latest driver and divactrl (possibly the SRPM from > Eicon). > > regards, > Armin > > On Tue, 16 Aug 2005 [EMAIL PROTECTED] wrote: > > Hello Armin, > > My setup is as follows: I have 1 bri with 2 spid's, or msn's. > > 2781980 and 2781984. > > > > If a call comes in to 2781980, and is active, and another call comes > > in to 2781980, the second call will be busy. > > > > A call to 278-1984 will proceed while the 1980 is busy. > > > > The telco tells me though that the bri should be capable of hunting > > on > > > it's own. > > > > I did this in the past with modem banks, but they were on top of > > centrex. > > > > What I would like to do is put an 800 number to point to the > > 278-1980, > > > and for the most part not use the 278-1984 except for maybe a disa. > > > > The eiconctrl monitor app is aware that the line is busy, and I do > > not > > > believe it is notifying asterisk of the issue. > > > > I am trying to move some lines to bri since my audio quality on pots > > has been horrible. The isdn is great, especially since you told me > > of > > > the ulaw modification I needed to make... I got lucky with this > > one, since they really could not install it without doing special > > construction, which I managed to avoid paying the big bucks for > > because the csr was nice about the 3 month delay. I set it up > > through > > > a panasonic dbs so the secretary can just hit a button, and I get > > immediate rings on 4 sip phones and my cell. I would love a PRI, > > but only need 4 channels max which is why I went with the bri. > > > > Compared to pots, the isdn is way better. I also find it much more > > stable than IP, to the point where it is worth the 1c/minute to use. > > > > Thanks for the help. > > > > Greg > > > > -Original Message- > > From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] > > Sent: Tuesday, August 16, 2005 2:06 AM > > To: Gregory Wiktor - ADCom Corp. > > Subject: RE: [Asterisk-Users] BRI Hunting, using both channels on > > one msn > > > > On ISDN, the second channel is automatically used if the first > > channel > > > is busy. > > Normaly you never get a busy signal, just because ONE channel is busy. > > Only if there is no application/phone available for
RE: [Asterisk-Users] system() app changed drastically! How do I useit now?
It would be prudent the test for success and continue rather than
failure and drop.
For example:
exten => s,5,GotoIf($["${SYSTEMSTATUS}" != "SUCCESS"]?105:6)
That way only the result that you know is good, Will continue a call..
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Jim Gottlieb
> Sent: Monday, September 26, 2005 10:22 PM
> To: asterisk-users@lists.digium.com
> Subject: Re: [Asterisk-Users] system() app changed
> drastically! How do I useit now?
>
> On 2005-09-26 at 18:15, Jim Gottlieb ([EMAIL PROTECTED]) wrote:
>
> > But since (as far as I know, without using AEL) there is no
> > conditional branching based on a variable, how am I
> supposed to use this?
>
> OK, I forgot about GotoIf. However, the doc is wrong (or at
> least incomplete), because it only mentions SUCCESS and
> FAILURE, but I'm finding SYSTEMSTATUS set to APPERROR.
>
> So I'm doing:
>
> exten => s,5,GotoIf($["${SYSTEMSTATUS}" = "APPERROR"]?105:6)
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[Asterisk-Users] re: DTMF woes, continued
Hi Yair, Please let me if you managed to fix the DTMF tone issue, which you were experiencing couple of months ago. If not can you share any advancement. I'm currently experiencing the same issue, I can make outbound calls but DTMF will not work when dialing IVRs. My configuration is [EMAIL PROTECTED] 1.5, registering to Voip provider (Symbio), codec is g.729 and dtmf mode is set to rfc2833. Your help will be much appreciated Thanks & Regards, Esteban _ View 1000s of pictures, profiles and more now at Lavalife http://lavalife.com.au ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] system() app changed drastically! How do I use it now?
On 2005-09-26 at 18:15, Jim Gottlieb ([EMAIL PROTECTED]) wrote:
> But since (as far as I know, without using AEL) there is no conditional
> branching based on a variable, how am I supposed to use this?
OK, I forgot about GotoIf. However, the doc is wrong (or at least
incomplete), because it only mentions SUCCESS and FAILURE, but I'm
finding SYSTEMSTATUS set to APPERROR.
So I'm doing:
exten => s,5,GotoIf($["${SYSTEMSTATUS}" = "APPERROR"]?105:6)
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RE: [Asterisk-Users] AsteriskJava - Queue
You may loose 'control' of the call but you can always 'get it back' Use the UnigueID of the call to track it throught Asterisk. You can palce a monitor event to redirect, bridge, drop, answer or antything else. > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > Sebastian Silva > Sent: Monday, September 26, 2005 6:33 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [Asterisk-Users] AsteriskJava - Queue > > Hi, I am using AsteriskJava and I have some problems, I will > appreciate any help... > > My system has the following architecture (in the server side): > > - An app server (connected to the asterisk console) > - An AGI Server (developed with AsteriskJava) > - An AGI Script (executed by the above AGI Server) > > In the client side (Agents answering call center calls): > > - A softphone > - A client program (used to search and register call details) > > Here is the thing: > > - From AGI Server I detect that a call is coming from PSTN > and launch the AGI Script > - From AGI Script I put the call in the queue and I loose the > control of the call (here is my first confusion) > - The agent answer the call (using his/her softphone) and I > get the event from the Asterisk Console with my App Server. > > Now, I need to play something (TTS, wav, etc) to the caller > based on the client application wich is connected to my App > Server. What I want you to know is that the information to be > played to the caller comes from an external source. > > So, my two big questions/confusions are: > > - How can I get the entire control of the call depending on > the status of the call, for example, if the call is in the > queue and I need to play or do something with it, where and > how I have the control? until now, when I put the call in the > queue I loss the control until the caller or the agent hangs the call. > > - Once the call is answered by the Agent, how can I unlink > the two channels (releasing the agent) to let the caller hear > the text that the agent sent. > > > Thanks in advance, > > Sebas > > > -- > Sebastian Silva > G R U P O G A U S S > Depto. Sistemas > Av. Libertador 6250 4 piso > Tl.: 4 706- (int. 121) > [EMAIL PROTECTED] > ___ > --Bandwidth and Colocation sponsored by Easynews.com -- > > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] system() app changed drastically! How do I use itnow?
Try the following:
exten => s,1,Answer
exten => s,2,Wait(1)
exten => s,3,Read(PIN,87)
exten => s,4,System(/usr/local/bin/pin -c ${PIN}) ; check it
exten => s,5,System(/usr/local/bin/pin -d ${PIN}) ; delete it
Exten => s,6,GotoIf($[${SYSTEMSTATUS} = FAILURE]?105:7)
exten => s,7,SetAccount(${PIN})
exten => s,8,Newt,pinout-config ; connect them
exten => s,105,Playback(5021); tell them their PIN is invalid
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Jim Gottlieb
> Sent: Monday, September 26, 2005 9:16 PM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] system() app changed drastically!
> How do I use itnow?
>
> We upgraded to the latest version of asterisk (because we
> needed some newer features), only to find all our PIN
> applications accepting any number the caller makes up!
>
> I traced this to the System application completely changing
> the way it deals with success or failure of the program it calls.
>
> Previously, if the PIN was completely bogus, we exited with
> -1, which caused asterisk to jump to priority n + 101 and we
> told the caller to take a hike. Now, instead it sets
> $SYSTEMSTATUS to either "SUCCESS" or "FAILURE".
>
> But since (as far as I know, without using AEL) there is no
> conditional branching based on a variable, how am I supposed
> to use this?
>
> I'd appreciate any ideas.
>
> Thank you.
>
>
> Here's an example of our one-time PIN setup.
> exten => s,1,Answer
> exten => s,2,Wait(1)
> exten => s,3,Read(PIN,87)
> exten => s,4,System(/usr/local/bin/pin -c ${PIN}) ; check it
> exten => s,5,System(/usr/local/bin/pin -d ${PIN}) ; delete it
> exten => s,6,SetAccount(${PIN}) exten =>
> s,7,Newt,pinout-config ; connect them
> exten => s,105,Playback(5021) ; tell them their PIN is invalid
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Re: R: [Asterisk-Users] Problem setting up TDM22B card
I know it seems basic but did you make sure and plug power into the board when you installed it into the PCI slot? I spent about three hours trying to get the dang thing to work in my machine until I decided to stick the card into another PCI slot. That is when I noticed that I had forgotten to ALSO plug power into the board from the power supply. Everything worked fine after that (yep, I was a noob). :-) Patrick ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] system() app changed drastically! How do I use it now?
We upgraded to the latest version of asterisk (because we needed some
newer features), only to find all our PIN applications accepting any
number the caller makes up!
I traced this to the System application completely changing the way it
deals with success or failure of the program it calls.
Previously, if the PIN was completely bogus, we exited with -1, which
caused asterisk to jump to priority n + 101 and we told the caller to
take a hike. Now, instead it sets $SYSTEMSTATUS to either "SUCCESS" or
"FAILURE".
But since (as far as I know, without using AEL) there is no conditional
branching based on a variable, how am I supposed to use this?
I'd appreciate any ideas.
Thank you.
Here's an example of our one-time PIN setup.
exten => s,1,Answer
exten => s,2,Wait(1)
exten => s,3,Read(PIN,87)
exten => s,4,System(/usr/local/bin/pin -c ${PIN}) ; check it
exten => s,5,System(/usr/local/bin/pin -d ${PIN}) ; delete it
exten => s,6,SetAccount(${PIN})
exten => s,7,Newt,pinout-config ; connect them
exten => s,105,Playback(5021); tell them their PIN is invalid
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[Asterisk-Users] Faxing via a sip extension with a digium e1 card
We are having problems with the above configuration. A Canon fax in a SPA3000, Asterisk 1.0.9 stable with single Digium E1 card. Fax's are received well but sending is next to impossible. Anyone seen or heard of any solutions to this type of issue? Regards Mark ACC Australia Phone: 02 9489 0544 Fax: 02 9489 0522 Mobile: 0401 286 686 www.accaustralia.com.au ___NOTICEThis electronic mail transmission contains confidential information intended only for the person(s) named. Any use, distribution, copying or disclosure by any other person is strictly prohibited. If you received this transmission in error, please notify the sender by reply e-mail and then destroy the message. Opinions, conclusions, and other information in this message that do not relate to the official business of ACC Australia shall be understood to be neither given nor endorsed by the Company. When addressed to ACC clients, any information contained in this e-mail is subject to the terms and conditions in the governing client contract. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Socket 478 Motherboard for use with TDM400P
I've used Biostar U8668D and Asus P4GE-MX. Chris - Original Message - From: "Brent August Torrenga" <[EMAIL PROTECTED]> To: Sent: Monday, September 26, 2005 6:09 PM Subject: [Asterisk-Users] Socket 478 Motherboard for use with TDM400P > Well, it has been a long saga over here trying to get a TDM400P to work > with an Intel D845HV motherboard, and the towel is close to being thrown > in. Does anyone know of a (few) motherboards that work well with the > TDM400P and are socket 478? > > --Brent > > ___ > --Bandwidth and Colocation sponsored by Easynews.com -- > > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TE110P Hanging up & sometimes not picking up on E&M T1
I'm getting unstable behavior with my newly installed TE110P T1 card. It
hangs up any incoming call anywhere from 20 seconds to 6 minutes.
Frequently, when you call back on our incoming T1 there'll be an
automated announcement (maybe from the telco?) stating "we are unable to
complete your call at this time". I simultaeously get a yellow alarm on
the system and shortly after the yellow alarm is cleared (appears the T1
card is reconnecting). Here's what I've tried:
* I'm using E&M signalling (not PRI), and have toyed with rxwink
times of 250ms to 450ms.
* I have also changed: busydetect=no, callprogress=no
There is also one other odd behavior that I'm noticing... within the
Asterisk CLI I'm seeing that when a call comes in the system appears to
be spawning several extensions at a time. For example, a single call
comes in and I see:
== Spawn extension (aa_2, s, 7) exited non-zero on 'Zap/19-1'
-- Executing Hangup("Zap/19-1", "") in new stack
== Spawn extension (aa_2, h, 1) exited non-zero on 'Zap/19-1'
== Spawn extension (aa_2, s, 7) exited non-zero on 'Zap/17-1'
== Spawn extension (aa_2, s, 7) exited non-zero on 'Zap/24-1'
== Spawn extension (aa_2, s, 7) exited non-zero on 'Zap/13-1'
== Spawn extension (aa_2, s, 7) exited non-zero on 'Zap/18-1'
== Spawn extension (aa_2, s, 7) exited non-zero on 'Zap/6-1'
To my untrained eye, it seems that Asterisk is over-reacting and
presenting the single incoming call with hoard of available channels.
Makes me think it has something to do with timing.
Any ideas on what's going on and suggestions on how to fix this?
thanks,
Sasch
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[Asterisk-Users] Socket 478 Motherboard for use with TDM400P
Well, it has been a long saga over here trying to get a TDM400P to work with an Intel D845HV motherboard, and the towel is close to being thrown in. Does anyone know of a (few) motherboards that work well with the TDM400P and are socket 478? --Brent ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] voipbuster advise
Greetings, 1.) Voipbuster does not support T.38. If you can get a clean connect using G.711u then the answer is maybe. Latency will ice a analog connection. 2.) That's built into the dialplan at VoipBuster. It's doubtful they'll remove the "routing charge" message, but you could always ask their customer service. :-) -Don -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of FaberK Sent: Monday, September 26, 2005 3:27 PM To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] voipbuster advise Hi, I'm using voipbuster at work, and I've got 2 questions: 1) Is it possible to send faxes using voipbuster connex? 2) Is it possible to cut off or cover the voice that say the charge per minute?(I've payed the '5' euro, and from that moment I've got it!). Of course I understand that is to let me know how much I'm going to spend, but I do not like it, expecially when I'm with clients. Any links, suggestions? Thanks -- .:FaberK:. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] voipbuster advise
Hi, I'm using voipbuster at work, and I've got 2 questions: 1) Is it possible to send faxes using voipbuster connex? 2) Is it possible to cut off or cover the voice that say the charge per minute?(I've payed the '5' euro, and from that moment I've got it!). Of course I understand that is to let me know how much I'm going to spend, but I do not like it, expecially when I'm with clients. Any links, suggestions? Thanks -- .:FaberK:. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CPU spiking with TDM400 cards fixed
Of possible interest to people having various issues with TDM400 cards, is that a fix has just been submitted to CVS for the issue where CPU usage would regularly spike up to 100% with the wctdm driver loaded. Regards, Richard ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FSX/UK analogue Phone rings all the time
On Mon, September 26, 2005 20:35, Asterisk said: > hi Asterisk users, > > I am in the UK and trying to get an asterisk system running. > > I have the SIP side of things running or limping along to the best of my > newbie > ability. > > I have a problem with a FXS card. Connecting a standard (Working) UK phone > makes > the phone ring all the time while on hook. Sounds like the A/B is being > coupled > onto the ring wire. I've heard somewhere that you need to connect the phone through a master socket. I think its something to do with the ringing signal from the FXS card. -- John ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] netappel
Hi guys, does anybody succesfully connect Asterisk with netappel? I've tryed using voipbuster settings, but doesn't work. Any suggestions, are wellcome. Thanks -- .:FaberK:. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Realtime.. : Unixodbc drivers
Hi! About realtime... Anybody knows a unixodbc driver for oracle (free or comertial)? I am working with a trial easysoft odbc-driver, but the commertial license is very expensive... Another question. res_odbc.so works with IODBC ? Regards. Jsalas ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] What ISDN hardware would you recommend?
Trying again... *Summary:* I need to have 2 machines with 4 BRI connections, 2 in NT mode, 2 in TE mode and 1 machines with 2 BRI connections, 1 in NT mode, 1 in TE mode; What card(s) should I put in to these servers? *The long story:* I have 3 locations I want to connect using (*) servers. 1 of those has a single BRI with a Siemens DECT PABX. 1 of those has two BRI's with 2 Siemens DECT PABX's, each serving a different area. 1 of those has two BRI's and a 2 port Nova Compact PABX with DECT First step would be to set up the (*) servers and have them interconnected. When all of that works we'd go on to connect them to the ISDN and connect the existing PABX's to the servers so we can - for now - maintain the existing environment but use (*) to route traffic on a least cost basis, as well as allow SIP/IAX connections from out of office locations. The machines themselves will not pose much of a problem, but what ISDN hardware would you recommend for this? (1 site with 1 TE and 1 NT mode port, 2 sites with 2 TE and 2 NT mode ports) TIA! -- Francesco Peeters GPG Key = AA69 E7C6 1D8A F148 160C D5C4 9943 6E38 D5E3 7704 If your program doesn't recognize my signature, please visit http://www.CAcert.org/index.php?id=3 to retrieve the Root CA certificate. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE405P V2 - Fantastic!
On Mon, Sep 26, 2005 at 07:17:36AM -0400, Andrew Kohlsmith wrote: > On Monday 26 September 2005 00:36, Kevin P. Fleming wrote: > > Bridged calls with 2nd gen firmware result in the audio never leaving > > the card; that's why you are seeing such an improvement. Essentially, > > the Zaptel 'native bridge' is pushed all the way down into the card, so > > the audio stream is never passed across the PCI bus (it's not even > > packetized, just directly connected between the two channels). > > This is why so many of us are pushing Digium to PLEASE FOR THE LOVE OF GOD > print a detailled list of what's improved with the new firmware... None of > us have any clear idea of what has changed from v1 to v2 and little things > like this are unbelievably important. > > Kind of like how some of us are also pushing for a more detailed changelog... > > not cvs log type of depth but Bugs fixed in this release: #105 #3033 #5050 > etc and features added/removed/changed with a little more detail. There was also an update to the echo canceller that went into the zaptel 1.0.x release branch at sometime that could be improving his echo problems. -- Matthew Fredrickson ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] AsteriskJava - Queue
Hi, I am using AsteriskJava and I have some problems, I will appreciate any help... My system has the following architecture (in the server side): - An app server (connected to the asterisk console) - An AGI Server (developed with AsteriskJava) - An AGI Script (executed by the above AGI Server) In the client side (Agents answering call center calls): - A softphone - A client program (used to search and register call details) Here is the thing: - From AGI Server I detect that a call is coming from PSTN and launch the AGI Script - From AGI Script I put the call in the queue and I loose the control of the call (here is my first confusion) - The agent answer the call (using his/her softphone) and I get the event from the Asterisk Console with my App Server. Now, I need to play something (TTS, wav, etc) to the caller based on the client application wich is connected to my App Server. What I want you to know is that the information to be played to the caller comes from an external source. So, my two big questions/confusions are: - How can I get the entire control of the call depending on the status of the call, for example, if the call is in the queue and I need to play or do something with it, where and how I have the control? until now, when I put the call in the queue I loss the control until the caller or the agent hangs the call. - Once the call is answered by the Agent, how can I unlink the two channels (releasing the agent) to let the caller hear the text that the agent sent. Thanks in advance, Sebas -- Sebastian Silva G R U P O G A U S S Depto. Sistemas Av. Libertador 6250 4 piso Tl.: 4 706- (int. 121) [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Polycom Setup Questions
OK I have just gone live with asterisk in a new office with approx 40 Polycom 501 handsets. I have a few questions: 1) Call Parking: I am able to park calls using the standard Asterisk call parking system (transfer to ext *70 etc...) I would like to make this a little easier for my users The Polycom 501s w/ SIP 1.5.2 seem to support some type of standard call parking, however I don't think it works with Asterisk. Is this true? Is there a way to integrate the to call parking system etc? 1a) If I can't use the Polycom built-in call park feature, is there a way to remap one of the buttons on the left (the services button for example) to dial *70 for my users? 2) Transferring Calls: They way our office operates, I would prefer the default transfer method to be a blind transfer. Is there a way to reprogram the Polycoms to default to blind transfers? There are more questions but that is all for now :-) Thanks, Matt ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ZapHFC Channel unavailable
I am running Asterisk 1.0.9 and the latest bristuff in combination with a
HFC isdn card, connected to a BRI interface.
For some reason, I am not able to have it dial out (see below). It exits
with DIALSTATUS=CHANUNAVAIL.
One thing that may be misconfigured is that it says: Signalling Type: PRI
Signalling.
How do I change this? Or is this the correct value?
I posted my configs below.
-- Console output: (note the PRI Signalling)
*CLI> zap show channel 1
Channel: 1
File Descriptor: 17
Span: 1
Extension:
Dialing: no
Context: default
Caller ID string:
Destroy: 0
InAlarm: 1
Signalling Type: PRI Signalling
Owner:
Real:
Callwait:
Threeway:
Confno: -1
Propagated Conference: -1
Real in conference: 0
DSP: no
Relax DTMF: yes
Dialing/CallwaitCAS: 0/0
Default law: alaw
Fax Handled: no
Pulse phone: no
Echo Cancellation: 128 taps, currently OFF
PRI Flags:
PRI Logical Span: Implicit
Actual Hookstate: Onhook
--
--
*CLI> bri debug span 1
Enabled debugging on span 1
*CLI> dial 99xx
*CLI> -- Executing Dial("OSS/dsp", "Zap/g1/xx|120") in new stack
Unable to create channel of type 'Zap'
== Everyone is busy/congested at this time
Exiting with DIALSTATUS=CHANUNAVAIL.
--
-- zapata.conf:
[channels]
switchtype = euroisdn
signalling = bri_cpe_ptmp
nationalprefix = 0
internationalprefix = 00
usecallingpres=yes
echocancel=yes
echocancelwhenbridged=yes
relaxdtmf=yes
rxgain=1.5
txgain=1.5
immediate=no
busydetect=no
callprogress=no
musiconhold=default
usecallerid=yes
callerid=asreceived
immediate=yes
signalling=bri_cpe_ptmp
group = 1
context=default
channel => 1-2
--
-- zaptel.conf:
span=1,1,3,ccs,ami
bchan=1,2
dchan=3
loadzone=nl
defaultzone=nl
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Re: [Asterisk-Users] Asterisk and Broadvoice, no incoming voice
Using voicepulse retail account because I need the caller ID name. Maybe they have added that to connect accounts by now. Jason Schafer wrote: This sounds like a winner, are you using voicepulse? Also, I downloaded the iso for AAH 1.5. Are there any noteworthy bugs that are fixed? I don't really like to upgrade unless I need to. Jason Paul wrote: connect.voicepulse.com allows up to 4 calls at a time coming into an $11/month DID with choice of IAX or SIP ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Grandstream 496 not working on cordless phone
Hi All, We are using SER/Asterisk, it works fine from X-lite to corded phones but have problems using a cordless phone on the Handytone 496. Has anyone experienced this problem ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk and Broadvoice, no incoming voice
Price is about the only good thing... quality? Jajajajaj reliable? Jajajajja -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jason Schafer Sent: Monday, September 26, 2005 3:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk and Broadvoice, no incoming voice I'm relatively new to the whole VOIP game, here's what I want to do. I am using VOIPJet for all of the outbound calls on our AAH box. I have one landline that I would like to busy forward to an inbound VOIP number. Broadvoice was recommended to me for price and quality. Can anyone make a suggestion for a good VOIP Provider for my inbound requirement? The bulk of my inbound calls will come in on the land line, but I would also like the leverage the group/conference feature in AAH (8+ext) and an inbound SIP seems to be a good answer for having a couple of different people call in at once (three people call the SIP number). Jason ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] how to connect two SIP channels
Hello, I have two extensions 100 and 200 and I need in "h" context dial first extension by asterisk and when the extensions 100 is answered, then to dial exten 200 and then bridge them together.. How can I do that? I need it to "call back" when called party is busy. thanks ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Command "sip show objects"
Hi, Why the command "sip show objects" isn't in the version 1.0.9? Exits any similar? Thanks. Ezequiel ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Areskicc LCR problem
I have got Areskicc installed with AMP and I can’t stress out how good and excellent this software is. I must say the author desire every credit for this software and I would like to say Thank you to Arezqui who wrote the software. Now there is one last thing before my whole asterisk calling card become prefect. And I am wondering whether anybody may know the answer of this. I created some rate card. And choose to do LCR. For a few routes I put cost to be 0 which I got those for free i.e I can call those country for free and then along side there is another rate card where my cost for those route is 0.01. Therefore when I choose LCR, I expect the Areskicc to choose the route for 0 and not the 0.01 route. But unfortunately it just love to spend my money and always go for the 0.01 unless I physically take the rate card off the tariff group. And I have tried to contact the areskicc support but no response. If anyone has this problem or got a bit of time please feel free to help me in any way .. Cheers Sam ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Performance tuning on dual Xeon EM64T and x86_64 Linux
Hi all, Just a couple of quick questions. I have a HP DL360 G4 (dual Xeon EM64T 3.0Ghz processors). I am using a TE411P in the system. 1. Should I run the a x86_64 Linux (CentOS) or just go with the plain old x86 version? Is there any benefit (or things to be aware of) on x86_64 vs x86? 2. This being a dual processor system, should I turn on or off hyper thrreading? Thanks in advance! ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and Broadvoice, no incoming voice
This sounds like a winner, are you using voicepulse? Also, I downloaded the iso for AAH 1.5. Are there any noteworthy bugs that are fixed? I don't really like to upgrade unless I need to. Jason Paul wrote: connect.voicepulse.com allows up to 4 calls at a time coming into an $11/month DID with choice of IAX or SIP ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Removing "-" (Dash) from Dialed Numbers
Nathan Pralle wrote: I am trying to enable dial-by-email by using LDAPget to query an Active Directory server. I've got it retrieving the phone number fine. Unforunately, the numbers stored in active directory are either in the format: (xxx) xxx- or xxx-xxx-. Is there any way to parse characters out of the dialed phone number so that I only end up with digits (remove spaces, parenthesis and dashes)? From there, my outbound routes can take care of where to send the call. Scott, This would be darned easy to do with the AGI and a perl script. $number=~s/-//g; $number=~s/ //g; $number=~s/\(//g; $number=~s/\)//g; change this to be... $number =~ s/\D//g; # \D is regexp for non-digit ...or if some letters were ok to dial (ABCD) you could do... $number =~ s/(?i)[^\dabcd]//g; # \d is regexp for digits # (?i) make the express case-insensitive # the [^...] construct effectively means # anything not specified between [ & ] my $two_cents = 'a grain of salt'; -- Christopher L. Wade, CCNA, CCDA, CQS-CIPTES, CQS-CWLSS ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Best drivers for HFC-S ISDN cards
Chris I have only ever used zaphfc drivers and for me they are perfect. Echo has never been a problem. It would be helpful if you were to provide a bit more information to the group about your configuration so we can try and help you work out the cause. Switching to capi or mISDN is unlikely to help and will almost certainly be a retrograde step as far as I hear from these forums. Best regards Tim Robinson Basingstoke, UK Chris Bagnall wrote: It seems that HFC-S cards can be connected with asterisk in a few different ways - isdn4linux, mISDN, chan_capi or zaphfc. I've tried isdn4linux (severe echo, reproducable on every inbound call) and zaphfc (intermittent echo, disappears within about 30 secs of the call starting). What's the recommended way to hook up these ISDN cards? Is switching to capi or mISDN likely to remove the echo problem completely, or is this one of those things one has to accept? Thanks in advance. Regards, Chris ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] change codec based on callerid (sip/iax)
trixter http://www.0xdecafbad.com wrote: On Mon, 2005-09-26 at 10:33 -0700, Michael D Schelin wrote: This can be done by modifying the source code. how helpful. If I modify it enough it will be 100% identical to windows xp, anything can be done by modifying any code. That however doesnt answer my question with anything that isnt obvious, such as is there a way to do it without a modification? Would the ser idea work (which may be better as the call volume would likely exceed asterisks ability to process calls anyway? We could set up an autoresponder that says "Modify the source code, search the wiki and rtfm" for every post to the list. As for the question. If ser can do it today, I would go that way. If ser can be running on the same box that's even better for some situations. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dialogic Cards Will they be available to NON AsteriskBE
Anyone know if the INTEL/Dialogic announcement will become available to us who do not use the asterisk BE? Just curious as I have used those cards in the past and they are VERY STABLE and VERY dependable with excellent quality. ~ron ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk + GNUGK + Asterisk-Addons ooh323
Brian C. Fertig wrote: yes.. I have looked. they are different. But when I unregister 1 the other will register.. Its only when I have 2 of them trying to register at the same time I have an issue. But yes the ID's are different in both of them. maybe you have the same aliases ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Removing "-" (Dash) from Dialed Numbers
I am trying to enable dial-by-email by using LDAPget to query an
Active Directory server. I've got it retrieving the phone number
fine. Unforunately, the numbers stored in active directory are
either in the format: (xxx) xxx- or xxx-xxx-. Is there
any way to parse characters out of the dialed phone number so that I
only end up with digits (remove spaces, parenthesis and dashes)?
From there, my outbound routes can take care of where to send the
call.
Scott,
This would be darned easy to do with the AGI and a perl script.
IE:
exten => _X.,1,agi,fixnumbers|${MyNumber}
exten => _X.,2,Dial(ZAP/g0/1${MyNumber})
Then, in a perl script called "fixnumbers" and inside the agi-bin directory:
## START CODE #
#!/usr/bin/perl -w
use strict;
use Asterisk::AGI;
$AGI = new Asterisk::AGI;
my %input = $AGI->ReadParse();
my $number=$ARGV[0];
$number=~s/-//g;
$number=~s/ //g;
$number=~s/\(//g;
$number=~s/\)//g;
print $AGI->set_variable('MyNumber',"$number");
exit;
### END CODE
Nathan
--
-
Nathan E. Pralle
Give the Director a Serpent Deflector
www.nathanpralle.com
-
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Re: [Asterisk-Users] Asterisk and Broadvoice, no incoming voice
connect.voicepulse.com allows up to 4 calls at a time coming into an $11/month DID with choice of IAX or SIP there was discussion here and other places before about broadvoice allowing the calls but then charging 3.9c a minute for the extra channels used shop carefully Jason Schafer wrote: I'm relatively new to the whole VOIP game, here's what I want to do. I am using VOIPJet for all of the outbound calls on our AAH box. I have one landline that I would like to busy forward to an inbound VOIP number. Broadvoice was recommended to me for price and quality. Can anyone make a suggestion for a good VOIP Provider for my inbound requirement? The bulk of my inbound calls will come in on the land line, but I would also like the leverage the group/conference feature in AAH (8+ext) and an inbound SIP seems to be a good answer for having a couple of different people call in at once (three people call the SIP number). Jason ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IBM x306 - some progress
Hi, I asked yesterday about a problem with x306 and IRQ sharing, didnt get much info, now, i was playing with lspci, and see something strange, lspci -v shows me the TDM400P card is on IRQ 7, and the SCSI card is also on IRQ 7, lspci -bv (from the man - b - shows "bus-centric view, as seen by the BUS and not by the kernel) shows me the TDM400P is on IRQ 5, why does the kernel puts it on IRQ 7 ? any insights much appriciated. Marco. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and Broadvoice, no incoming voice
IMHO - you should not use price and quality in the same sentence for BV. On Mon, 2005-09-26 at 15:20 -0400, Jason Schafer wrote: > I'm relatively new to the whole VOIP game, here's what I want to do. I > am using VOIPJet for all of the outbound calls on our AAH box. I have > one landline that I would like to busy forward to an inbound VOIP > number. Broadvoice was recommended to me for price and quality. > > Can anyone make a suggestion for a good VOIP Provider for my inbound > requirement? The bulk of my inbound calls will come in on the land > line, but I would also like the leverage the group/conference feature in > AAH (8+ext) and an inbound SIP seems to be a good answer for having a > couple of different people call in at once (three people call the SIP > number). > > Jason > ___ > --Bandwidth and Colocation sponsored by Easynews.com -- > > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Call Back On Busy?
Not a bad idea, thank you for that. I'll look into
it
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of BJ
WeschkeSent: Monday, September 26, 2005 2:37 PMTo:
Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re:
[Asterisk-Users] Call Back On Busy?
Is there a functional reason why you'd use MeetMe here? I
think probably the easiest way to accomplish this is to use an DeadAGI script
which can be invoked via the 'h' extension in the context that would then
perform the functionality you're looking for and if they get through it should
just bridge the original caller back in.
On 9/26/05, Sherwood
McGowan <[EMAIL PROTECTED]>
wrote:
Anyone
else out there have some thoughts? The customer wants to be able to control
what can be redialed on busy, such as no international. I'm having my doubts
as to whether or not this can be done. My idea seems like it would work, but
after the customer hangs up, wouldn't the context stop processing?
Thanks,
SKM
From: [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED]] On Behalf Of
Damon EstepSent: Monday, September 26, 2005 10:15 AM To: Asterisk Users Mailing List - Non-Commercial
DiscussionSubject: RE: [Asterisk-Users] Call Back On
Busy?
This may not
apply to your situation, but many ATAs and SIP phones have this feature
built in to the device.
We use
Linksys/Sipura and auto redial and last call return work without any
special setup.
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] ] On Behalf Of Sherwood
McGowanSent: Monday,
September 26, 2005 7:45 AMTo: 'Asterisk Users Mailing List -
Non-Commercial Discussion' Subject: [Asterisk-Users] Call Back
On Busy?
I know it's been touched on
before, but no answers have been found to the best of my knowledge. I'm
using a SIP only setup, with a sip provider giving PSTN and would like to
see if anyone has an idea for creating redial busy using ${DIALSTATUS} and
possibly MeetMe?
I figure something like this,
but want to get feedback
1. Get callers last dialed
number, if international number, do not allow.
2. Playback a stuttertone to
caller
3. Disconnect
caller
4. Ring intended party check
dial status. If busy, wait 120 seconds and try again (do this for a
total of 15 minutes)
5. If it's picked up, playback
an announcement to the party and put them in a meetme
conference
6. Ring the original caller
and bridge them to the meetme conference.
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Re: [Asterisk-Users] change codec based on callerid (sip/iax)
On Mon, 2005-09-26 at 10:33 -0700, Michael D Schelin wrote: > This can be done by modifying the source code. > how helpful. If I modify it enough it will be 100% identical to windows xp, anything can be done by modifying any code. That however doesnt answer my question with anything that isnt obvious, such as is there a way to do it without a modification? Would the ser idea work (which may be better as the call volume would likely exceed asterisks ability to process calls anyway? -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FSX/UK analogue Phone rings all the time
hi Asterisk users, I am in the UK and trying to get an asterisk system running. I have the SIP side of things running or limping along to the best of my newbie ability. I have a problem with a FXS card. Connecting a standard (Working) UK phone makes the phone ring all the time while on hook. Sounds like the A/B is being coupled onto the ring wire. I plugged the phone into a RJ11/BT6312 adapter but it sounds very like like cross wiring somehow. Has anyone else come across this as I need this port to connect to a conventional PABX. Thanks Gary ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE405P V2 - Fantastic!
Andrew Kohlsmith wrote: This is why so many of us are pushing Digium to PLEASE FOR THE LOVE OF GOD print a detailled list of what's improved with the new firmware... None of us have any clear idea of what has changed from v1 to v2 and little things like this are unbelievably important. The list for the 2nd gen firmware has been posted to this mailing list already, IIRC. It includes: - support for VPM400M - more efficient bus-master DMA operations - on-card bridging of channels (timeslot interchange) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and Broadvoice, no incoming voice
I'm relatively new to the whole VOIP game, here's what I want to do. I am using VOIPJet for all of the outbound calls on our AAH box. I have one landline that I would like to busy forward to an inbound VOIP number. Broadvoice was recommended to me for price and quality. Can anyone make a suggestion for a good VOIP Provider for my inbound requirement? The bulk of my inbound calls will come in on the land line, but I would also like the leverage the group/conference feature in AAH (8+ext) and an inbound SIP seems to be a good answer for having a couple of different people call in at once (three people call the SIP number). Jason ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Carrier Access - Access Bank I config
I tried for weeks with an AB I, and never got anywhere...I could not get the T1 to sync properly. I switched exclusively to ADIT 600's and have had no issues since. -Darren From: [EMAIL PROTECTED] on behalf of Time Bandit Sent: Mon 9/26/2005 2:25 PM To: asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Carrier Access - Access Bank I config Hi, Is there somebody using an Access Bank I with Asterisk that could share the secret ingredients needed to make it work ? I've searched around and found some info, I tryed almost every configuration possible but I can't seem to find the right combination. If someone could provide me with the config needed on Asterisk as well as the dip-switch settings on the channel bank part, I would be really greatfull. Thanks ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users <>___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Removing "-" (Dash) from Dialed Numbers
That solutions kind of works, but only for
phone numbers that are in the format xxx-xxx-. The records I am receiving
from LDAP vary significantly because the user can control their entry. So I
can receiving mostly the following formats…
(xxx) xxx-
xxx xxx-
xxx-xxx-
Ideally, I’d like a function that
would remove all characters but the numbers, but I haven’t found a way to
do that. If I were more fluent in C, I might write a module to do it. Instead,
I added some extensions to try to compensate for the common formats, and it
looks like this…
[app-clean-ldap]
exten =>
_XXX-XXX-,1,SetVar(AreaCode=${EXTEN:0:3})
exten =>
_XXX-XXX-,2,SetVar(Prefix=${EXTEN:4:3})
exten =>
_XXX-XXX-,3,SetVar(Suffix=${EXTEN:8:4})
exten =>
_XXX-XXX-,4,Goto(from-internal,91${AreaCode}${Prefix}${Suffix},1)
exten => _(XXX)
XXX-,1,SetVar(AreaCode=${EXTEN:1:4})
exten => _(XXX)
XXX-,2,SetVar(Prefix=${EXTEN:5:3})
exten => _(XXX)
XXX-,3,SetVar(Suffix=${EXTEN:9:4})
exten => _(XXX)
XXX-,4,Goto(from-internal,${AreaCode}${Prefix}${Suffix},1)
exten => _XXX
XXX-,1,SetVar(AreaCode=${EXTEN:0:3})
exten => _XXX
XXX-,2,SetVar(Prefix=${EXTEN:4:3})
exten => _XXX
XXX-,3,SetVar(Suffix=${EXTEN:8:4})
exten => _XXX
XXX-,4,Goto(from-internal,91${AreaCode}${Prefix}${Suffix},1)
My only problem now is that I can’t
pattern match a space character in the last two sections. I was able use a
period as a wildcard to match the space, but that returns unexpected results if
someone has more than one character in there. Does anyone know how to match a
space character in the dial plan? Help is always appreciated.
Thanks,
Scott Miller
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Colin Anderson
Sent: Friday, September 23, 2005
2:47 PM
To: 'Asterisk
Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users]
Removing "-" (Dash) from Dialed Numbers
Not very elegant, but:
exten =>
1,1,SetVar(MyNumber=780-555-1212)
exten =>
1,2,SetVar(AreaCode=${MyNumber:3:3})
exten =>
1,3,SetVar(Prefix=${MyNumber:5:7})
exten =>
1,4,SetVar(Suffix=${MyNumber:8:11})
exten =>
1,5,Dial(ZAP/g0/1${AreaCode}${Prefix}${Suffix})
hth
-Original Message-
From: Scott Miller
[mailto:[EMAIL PROTECTED]
Sent: Friday, September 23, 2005
1:22 PM
To:
asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Removing
"-" (Dash) from Dialed Numbers
I am trying to enable dial-by-email by using LDAPget to
query an Active Directory server. I've got it retrieving the phone number
fine. Unforunately, the numbers stored in active directory are either in
the format: (xxx) xxx- or xxx-xxx-. Is there any way
to parse characters out of the dialed phone number so that I only end up with
digits (remove spaces, parenthesis and dashes)? From there, my
outbound routes can take care of where to send the call.
Help is always appreciated! J
Thank you,
Scott Miller
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RE: [Asterisk-Users] Need Help on Areski Calling Card Solution plz
Uppal, Can you guide me through in seting up trunk, ratecard etc. Goksie From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Junaid Uppal Sent: Sunday, September 25, 2005 12:19 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Need Help on Areski Calling Card Solution plz AreskiCC works great for me , i've been using it for ~ 500 + cards scene and it works awesome for me! really , the guy did a REALLY good job , trust me. cheers ~uppal On 9/25/05, chawki hammoud <[EMAIL PROTECTED]> wrote: Hi: My experience with Areski is I wasn't able to get it to work and wasn't able to get help including from the owner of "idiot guide" who inturns wasn't able to get areski to work either according to him. I easily downloaded astcc and works fine Regards; Chawki Hammoud --- ADEGOKE ARUNA < [EMAIL PROTECTED]> wrote: > Can someone share its working files experience on > areskicc with me. > > I got it installed but my sip user and iax could not > get registered talkless > of making call and all the include directives > instructed in the idiot guide > were followed. > > Can someone share its experience with me on this? > > Aruna > > -Original Message- > From: [EMAIL PROTECTED] > [mailto: [EMAIL PROTECTED]] On > Behalf Of CM Rahman Jr. > Sent: Tuesday, July 19, 2005 8:50 PM > To: Asterisk Users Mailing List - Non-Commercial > Discussion > Subject: Re: [Asterisk-Users] Comments on Areski > Calling Card Solution plz > > > I am using it. I liked it. The guy did a good job. > He doesn't have the agent > > module yet. But I think that is on its way. > > Thanks > > Quoting Arnd Vehling <[EMAIL PROTECTED]>: > > > Hi, > > > > can anyone who has the Areski Calling Card > solution on Asterisk > > working comment on it? Is is stable enough for a > production system? > > Any pros and cons? > > > > thx, > > > >Arnd > > > > > > ___ > > 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 > > > > > CM Rahman Jr. > CTO > CCS Internet > www.ccsi.com > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ > --Bandwidth and Colocation sponsored by Easynews.com > -- > > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk SMS and sprintpcs
Well, you need to be a bit more specific. How are you trying to send it? Are you using an SMSC? What kind of lines do you have? -- Tom On 9/26/05, Jerry Geis <[EMAIL PROTECTED]> wrote: > Does anyone know about sending SMS messages to a sprint pcs phone. > > Can you give me a few details. Thanks, > > Jerry > > ___ > --Bandwidth and Colocation sponsored by Easynews.com -- > > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- Tom ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ZAP ISDN losing digits
I had similar problem using Digium TE406 card. Try update the driver. I worked for me. Good luck. AK On 9/23/05, maka <[EMAIL PROTECTED]> wrote: Hi all, I got into a strange problem here. I've got an asterisk box with bristuff-0.2.0-RC7k, and a HFC PCI ISDN card, running in NT mode. The ISDN card is connected to a S0 bus and to a Siemens ISDN PBX. Two phones are connected to the ISDN PBX and are successfully getting calls from the asterisk box. When dialling from one of the phones, the ZAP channel seems to be missing out on some of the dialled digits everytime, i.e. if I dial 099557896, the asterisk box receives 09955896 sometimes, or 0995789, or something like that. This only happens on one of the phones, the other one is dialling fine and digits are being recognized well. I already tried setting relaxdtmf=yes in zapata.conf, but to no effect. If anyone has any idea what might be wrong, appreciate the feedback..Cheers ___ --Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing list Asterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Extension availabilty
more information about the hint priority in the extension file: http://www.voip-info.org/tiki-index.php?page=Asterisk%20standard%20extensions http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20BristuffDevstate > I have a client that has an old Merlin system. They would like to move to an > Asterisk based system, however, with their existing system each phone is > capable of displaying who is on the phone within there office. This is done > by lighting a red light for each line(extension) that is in use. Has anyone > been able to neatly create this feature? Perhaps an XML application can be > written for the Cisco 7960's that would be capable of displaying which > extension is being used and which extensions are not in use. Any suggestions > would be appreciated. > > > Thanks in advance, > -Josh > ___ > --Bandwidth and Colocation sponsored by Easynews.com -- > > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Extension availabilty
I have had good luck with a php-based status webpage. My users don't mind having a page up in their browser. If you were to use Soundpoint IP600s, this dynamic page could be shown on the phone, which IMO would be great unless you had lots of users to monitor. Also, the polycom soundpoint line of phones use SIP PRESENCE messages to show 'buddy' status. This works pretty well most of the time but I'm sure it can only get better over time. Joshua Laroff wrote: I have a client that has an old Merlin system. They would like to move to an Asterisk based system, however, with their existing system each phone is capable of displaying who is on the phone within there office. This is done by lighting a red light for each line(extension) that is in use. Has anyone been able to neatly create this feature? Perhaps an XML application can be written for the Cisco 7960's that would be capable of displaying which extension is being used and which extensions are not in use. Any suggestions would be appreciated. Thanks in advance, -Josh ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mojo <[EMAIL PROTECTED]> Office Manger, Horan & Company, LLC (907) 747- x112 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] didgium card in india
Capt MS wrote: thanks for the reply Is Digium card compatible with EPABX standards available in india , further how much does a card with three FXS and one FXO interface cost, Do u have any experience of implenting the same , I am in army what we lookin at is voice gateway to interface our PBX with the data network so that we have one underlying network to handle , any suggestions on how to implement in a cost effective manner. I am using Digium card in India (Trivandrum, Kerala) for a small call center application. What I did was to purchase the card in US, send it across to my friend in his US address and he brought it along when he came, but I guess this option is not applicable to you. 3 FXS and 1 FSO will cost some thing under Rs. 15,000, with out duty. See here for exact prices. http://store.yahoo.com/asteriskpbx/noname.html I tried it here with BSNL and a Siemens PBX, I am not receiving the callerid and it does not detect remote hangup. Pl mail me offline if you need further information. regards, raj ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CAS Question
I have to replace a custom PBX, that is infront on a IVR system based on OLD NMS AG-E1 Card. The Cards is configurated with CAS Digitalmode, someone can give me some info about Digim Cards CAS configuration i need a conversion Table? I wanto to don't touch configuration on winbox, i want only replace HWPBX box with asterisk. Diagram Telco E1 ===>Proprietary PBX(CAS)===>IVR Server AG-E1 Regards _ Get free infected, boring, wrong, empty, or any other email for yourself. Go to ---> http://www.mailchoose.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Back On Busy?
Is there a functional reason why you'd use MeetMe here? I think probably the easiest way to accomplish this is to use an DeadAGI script which can be invoked via the 'h' extension in the context that would then perform the functionality you're looking for and if they get through it should just bridge the original caller back in.
On 9/26/05, Sherwood McGowan <[EMAIL PROTECTED]> wrote:
Anyone else out there have some thoughts? The customer wants to be able to control what can be redialed on busy, such as no international. I'm having my doubts as to whether or not this can be done. My idea seems like it would work, but after the customer hangs up, wouldn't the context stop processing?
Thanks,
SKM
From: [EMAIL PROTECTED] [mailto:
[EMAIL PROTECTED]] On Behalf Of Damon EstepSent: Monday, September 26, 2005 10:15 AM
To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [Asterisk-Users] Call Back On Busy?
This may not apply to your situation, but many ATAs and SIP phones have this feature built in to the device.
We use Linksys/Sipura and auto redial and last call return work without any special setup.
From:
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
] On Behalf Of Sherwood McGowanSent: Monday, September 26, 2005 7:45 AMTo: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] Call Back On Busy?
I know it's been touched on before, but no answers have been found to the best of my knowledge. I'm using a SIP only setup, with a sip provider giving PSTN and would like to see if anyone has an idea for creating redial busy using ${DIALSTATUS} and possibly MeetMe?
I figure something like this, but want to get feedback
1. Get callers last dialed number, if international number, do not allow.
2. Playback a stuttertone to caller
3. Disconnect caller
4. Ring intended party check dial status. If busy, wait 120 seconds and try again (do this for a total of 15 minutes)
5. If it's picked up, playback an announcement to the party and put them in a meetme conference
6. Ring the original caller and bridge them to the meetme conference.
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Re: [Asterisk-Users] Extension availabilty
On Snom phones this feature works (look at the "Hint" Command in extension.conf.) Support for this should come for the Grandstream GXP2000, currently it does not working. Cisco 79x0, i dont know. > I have a client that has an old Merlin system. They would like to move to an > Asterisk based system, however, with their existing system each phone is > capable of displaying who is on the phone within there office. This is done > by lighting a red light for each line(extension) that is in use. Has anyone > been able to neatly create this feature? Perhaps an XML application can be > written for the Cisco 7960's that would be capable of displaying which > extension is being used and which extensions are not in use. Any suggestions > would be appreciated. > > > Thanks in advance, > -Josh > ___ > --Bandwidth and Colocation sponsored by Easynews.com -- > > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Extension availabilty
FOP does this quite nicely From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joshua LaroffSent: Monday, September 26, 2005 1:57 PMTo: Asterisk-Users@lists.digium.comSubject: [Asterisk-Users] Extension availabilty I have a client that has an old Merlin system. They would like to move to an Asterisk based system, however, with their existing system each phone is capable of displaying who is on the phone within there office. This is done by lighting a red light for each line(extension) that is in use. Has anyone been able to neatly create this feature? Perhaps an XML application can be written for the Cisco 7960's that would be capable of displaying which extension is being used and which extensions are not in use. Any suggestions would be appreciated.Thanks in advance,-Josh ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and Broadvoice, no incoming voice
Darren Wright wrote: I am also a long time client, and have no incoming BV today. -Darren it works here today but they can be a bit unpredictable I use a cheap byod lite account mostly as a test tool. I figure if they grow up someday I might use them more. I have been wondering if they will meet the FCC deadlines or just fade away. At least some providers have been sending notices and collecting street addresses last few months. Others look like they are not really preparing to stay in the business when the deadlines hit. Maybe they are hoping another provider will buy the customer base and DID's? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Extension availabilty
The snom360 phones along with the current CVS-HEAD of Asterisk can presently do this. You'll want to do a wiki search on "Hint" in the dialplan for implementation details. Polycom has also just released a DSS sidecar to go with their 601 model phones, but the firmware to support more than 8 appearances at a time is still in the works. If you need something "now", I'd go with snom360's and Asterisk. I have deployed this already in production and it is working quite well. The DSS LED lights solid when the person is on the line, and blinks when their phone is ringing with an incoming call. On 9/26/05, Joshua Laroff <[EMAIL PROTECTED]> wrote: I have a client that has an old Merlin system. They would like to move to an Asterisk based system, however, with their existing system each phone is capable of displaying who is on the phone within there office. This is done by lighting a red light for each line(extension) that is in use. Has anyone been able to neatly create this feature? Perhaps an XML application can be written for the Cisco 7960's that would be capable of displaying which extension is being used and which extensions are not in use. Any suggestions would be appreciated. Thanks in advance,-Josh___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Ring requested on channel already in use
I posted this 1.2.0-beta1 success story to asterisk-dev, and someone recommended that asterisk-users might benefit from it as well. Thanks, Alan Ferrency pair Networks, Inc. [EMAIL PROTECTED] -- Forwarded message -- Date: Thu, 22 Sep 2005 17:35:08 -0400 (EDT) Subject: [Asterisk-Dev] Re: Ring requested on channel already in use To: asterisk-dev@lists.digium.com > alan wrote: > > A problem was recently posted on the Asterisk-Users mailing list, and it > > went unresolved. Now that it's plaguing our production system as well, I > > need to look into it further. > Good report, lots of information. See if you can reproduce it in CVS-HEAD > (Asterisk, libpri, zaptel) > You need to test this with cvs head (1.2beta) first to see if it's not > already fixed... I am happy to say that since we upgraded to 1.2.0-beta1, our problems with Asterisk instability have not recurred. Our uptime is over a week, with the last restart a result of the upgrade. Thanks! I didn't like to see the answer "upgrade your production system to a beta version," but the truth is, it was working poorly enough that it was basically impossible not to at least try it. Here is a summary of the symptoms we were seeing in 1.0.9, for others with this issue who may benefit from an upgrade: We narrowed the problem down to this sequence of events: - an incoming Zap call on a PRI channel - was sent to the queue - and answered by a AgentCallbackLogin queue agent - who was using a SIP phone - and the agent attempted to SIP REFER transfer the call - to another AgentCallbackLogin agent on a SIP phone That's a lot of channels (zap -> agent -> local -> sip, transferring to agent -> local -> sip). When this happened, we saw these symptoms: - Rarely, the transfer succeeded. - More often, the ZAP channel was put in limbo and both SIP parties were dropped; or the transfer completed but there was one-way audio from Zap to SIP only. - Often, when the transfer failed, Asterisk was left in an inconsistent state, and would not function correctly until a restart was performed. -- asterisk -r consoles could not execute commands successfully -- "sip show channels" produced bogus output -- incoming Zap calls (over a PRI) resulted in "Ring requested on channel... already in use" errors, and the calling party was dropped immediately. After this experience with 1.2, I'd say that the upgrade should not cause many problems, as long as you thoroughly research and implement all required configuration changes. We have not experienced any problems with 1.2 which weren't also problems in 1.0.8/9, but we have had many other little issues solved which we were previously trying to ignore. Thank you very much, Alan Ferrency pair Networks, Inc. [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: sipuras 841 bad sound
> Re: sipuras 841 bad sound (Juan Jose Comellas) > On Tuesday 20 September 2005 20:46, Anton Krall wrote: > > I have a problems with some sipuras 841 and asterisk 1.0.9. > > (upgrade the firmware was suggested and completed, and didn't fix the > problem.) There are a few little configuration details which are hard to catch on the SPA-841, which can affect sound quality. * RTP packet size: 0.20 On the "SIP" tab of the Advanced Admin page, the "RTP packet size" is shown, measured in seconds. It defaults to 0.03, however Asterisk is hardcoded to use 0.02. This mismatch can cause sound issues. * Silence Supp Enable: Off On the "Ext1" and "Ext2" tabs of Advanced Admin, the "Silence Supp Enable" option must be turned off. This is Silence Suppression, which causes the phone to stop sending RTP packets when the phone detects silence in the handset. Asterisk 1.0.9 does not support silence suppression, so this option must be turned off, or audio stream timing will fail a lot. We have a bunch of SPA-841's in service, and we're just finishing working out the bugs in the system. Our latest audio issue, as far as we can tell, was caused by a Duplex Mismatch between the ethernet port on the Asterisk server, and the ethernet port on the switch it was connected to. When one is set to full duplex and the other half duplex, you get random, intermittant periods of massive packet loss/jitter, which messes up audio something fierce. I've found http://www.voiptroubleshooter.com/ to be a good source of info on diagnosing random "audio is bad" issues. It has sound clips of the different kinds of "audio is bad" problems, along with info on what might cause that kind of problem. Alan ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Carrier Access - Access Bank I config
Hi, Is there somebody using an Access Bank I with Asterisk that could share the secret ingredients needed to make it work ? I've searched around and found some info, I tryed almost every configuration possible but I can't seem to find the right combination. If someone could provide me with the config needed on Asterisk as well as the dip-switch settings on the channel bank part, I would be really greatfull. Thanks ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: VOIP in Japan using Freebit
Have you tried: [EMAIL PROTECTED]:[EMAIL PROTECTED] [EMAIL PROTECTED]:mypass:[EMAIL PROTECTED] [EMAIL PROTECTED]:mypass:[EMAIL PROTECTED]/05075034132 ? Sometimes SIP providers require the realm in the username, so the first part should have the @blah Then, the third part, is the callerid so it shouldn;t be required, and the last part, is the extension notification or something like that, I never use it. Always include the pass. Regards! Alchaemist "Pikoro" <[EMAIL PROTECTED]> wrote in message news:[EMAIL PROTECTED] > Has anyone had any experience using a VOIP provider in Japan? > > No matter what I try, my REGISTER string kicks back one of 2 errors: > Got SIP response 481 "Call/Transaction Does Not Exist" back from x.x.x.x > or > Got SIP response 400 "Bad Request" back from x.x.x.x > > My register string is as follows: > [EMAIL PROTECTED] > > I have tried the following also: > 05075034132:[EMAIL PROTECTED] > [EMAIL PROTECTED]/05075034132 > 05075034132:[EMAIL PROTECTED]/05075034132 > myuserid:[EMAIL PROTECTED] > > and variations of the above. > > Is there any other information I could provide in order to get some help? > > I guess another thing I am looking for is a list of possible registration > strings.. I'll try them all :D > > Cheers > > ___ > --Bandwidth and Colocation sponsored by Easynews.com -- > > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk SMS and sprintpcs
Does anyone know about sending SMS messages to a sprint pcs phone. Can you give me a few details. Thanks, Jerry ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] WRT54GP2 SIP server on LAN port
Thanks for the information Sherwood.
Then the question I had if the normal routing works for the SIP proxy
works with a LAN server.
But I cant get a success in connecting the router LINE1 to Asterisk.
WRT54GP2 says as status "Can't connect to login server" and there is no
connection attempt when running "sip debug" with "verbose 4".
In my sip.conf this is specified:
[linksys]
type=friend
host=dynamic
username=100
secret=x
canreinvites=no
context=outgoing-sip
And in extensions.conf
[default]
exten => s,1,Dial(SIP/linksys|30|gr)
exten => s,2,VoiceMail(u100)
exten => s,3,Congestion
[outgoing-sip]
exten => _[0-9#*].,1,Dial(SIP/blixtvik-sip/${EXTEN}||t)
Now incoming calls gets the following loggs:
-- Executing Dial("SIP/0755xx-5499", "SIP/linksys|30|gr") in new
stack
Sep 26 19:55:34 NOTICE[5525]: app_dial.c:777 dial_exec: Unable to create
channel of type 'SIP'
== Everyone is busy/congested at this time
-- Executing VoiceMail("SIP/0755xxx-5499", "u100") in new stack
-- Playing 'vm-theperson' (language 'se')
-- Playing 'digits/1' (language 'se')
== Spawn extension (default, s, 2) exited non-zero on
'SIP/0755xxx-5499'
Sep 26 19:55:37 ERROR[5525]: cdr_sqlite.c:136 sqlite_log: cdr_sqlite:
attempt to write a readonly database
Sep 26 19:55:37 ERROR[5525]: cdr_csv.c:222 csv_log: Unable to re-open
master file /var/log/asterisk//cdr-csv//Master.csv : Permission denied
The answering machine works but it will not get connected with my WRT54GP2.
See anything that causes WRT54GP2 not to be able to register to Asterisk?
~Johannes
> Actually, just point the line you want to use to a local ip address (the
> asterisk server). I currently do this with my service. i.e. If your
> Asterisk
> server is 192.168.15.200, just make the proxy for line 1 that address. It
> routes internally just fine.
>
> Sherwood McGowan
>
>
>
> _
>
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Tom Vile
> Sent: Sunday, September 25, 2005 5:45 PM
> To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
> Non-Commercial
> Discussion
> Subject: Re: [Asterisk-Users] WRT54GP2 SIP server on LAN port
>
>
> what I do is loopback the WAN port to a LAN port and am able to use both
> (ie) take a cable from the wan port of the router and plug it into the lan
> port on the same router. This will give you a local ip and it still
> should
> allow connection out to your other provider.
>
>
> On 9/25/05, Johannes <[EMAIL PROTECTED]
> > wrote:
>
> Hi,
>
> I'm trying to set up Asterisk behind my WRT54GP2 router that has a
> intergrated ATA box.
> My box are not locked in any way so I can access and change all settings.
>
> Now to the problem...
> I have gotten Asterisk to register with my provider and everything works
> just well..
> Now it's time to get the intergrated ATA to connect to asterisk.
> But the asterisk box in located on the LAN ports of the WRT54GP2.
> I can't get the router to connect to Asterisk.
>
> The question is then if the router does not use the normal routing table
> and will force the connect to the SIP gateway to the WAN port even that I
> specified a LAN IP as the gateway.
>
> Has anyone set up the WRT54GP2 to connect to a asterisk server thats on
> the LAN ports with a LAN IP? Or is this impossible?
>
> Regards,
> ~Johannes
>
> ___
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>
> 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
>
>
>
>
>
> --
> Tom Vile
> Baldwin Technology Solutions, Inc
> Consulting - Web Design - VoIP Telephony
> www.baldwintechsolutions.com
> Phone: 518-631-2855 x205
> Fax: 518-631-2856
>
> ___
> --Bandwidth and Colocation sponsored by Easynews.com --
>
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
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RE: [Asterisk-Users] Call Back On Busy?
Anyone else out there have some thoughts? The customer
wants to be able to control what can be redialed on busy, such as no
international. I'm having my doubts as to whether or not this can be done. My
idea seems like it would work, but after the customer hangs up, wouldn't the
context stop processing?
Thanks,
SKM
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Damon
EstepSent: Monday, September 26, 2005 10:15 AMTo:
Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE:
[Asterisk-Users] Call Back On Busy?
This may not apply to
your situation, but many ATAs and SIP phones have this feature built in to the
device.
We use Linksys/Sipura
and auto redial and last call return work without any special
setup.
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sherwood McGowanSent: Monday, September 26, 2005 7:45
AMTo: 'Asterisk Users Mailing List - Non-Commercial
Discussion'Subject: [Asterisk-Users] Call Back On
Busy?
I know it's been touched on
before, but no answers have been found to the best of my knowledge. I'm using
a SIP only setup, with a sip provider giving PSTN and would like to see if
anyone has an idea for creating redial busy using ${DIALSTATUS} and possibly
MeetMe?
I figure something like this, but
want to get feedback
1. Get callers last dialed number,
if international number, do not allow.
2. Playback a stuttertone to
caller
3. Disconnect
caller
4. Ring intended party check dial
status. If busy, wait 120 seconds and try again (do this for a total of
15 minutes)
5. If it's picked up, playback an
announcement to the party and put them in a meetme
conference
6. Ring the original caller and
bridge them to the meetme conference.
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[Asterisk-Users] Re: Message Waiting Indicator (MWI) for remote voice mail?
I haven't received any responses. Just wanted to follow up and see if anyone has ideas? It seems like there ought to be a way to do this, especially since the TDM400 FXS card is able to send the proper signal to the connected phone. It seems like there just needs to be a way to configure the FXO card to pass though or bridge that signal/information to the FXS card when it is received at the FXO card. VOIP VM -> Sipura -> Phone worked. Asterisk VM -> FXS -> Phone works. Just need a way to do: VOIP VM -> Sipura -> FXO -> ??? -> FXS -> Phone The above works great for everything else I've tried so far except for passing through the message waiting indicator. Thanks for any ideas! On 9/24/05, Brian McEntire <[EMAIL PROTECTED]> wrote: I have Asterisk voice mail setup locally. It works great, I'm impressed! Some details about my system: I'm using a TDM22B card to interface with both the PSTN and a VOIP provider. I'm running 1.2-beta from CVS. I have a regular VTech phone plugged into one of the FXS ports. Asterisk is able to indicate when a local voicemail message is waiting via the LCD display of my analog phone. It also gives a broken dial tone. This is achieved by specifying mailbox= in zapata.conf and possibly also by specifing adsi=yes in the same file. The question I have is this: I also have voicemail with my VOIP provider. Before jumping into Asterisk, the VOIP provider could send the message waiting indicator to my phone when I had new messages. After putting Asterisk between my analog handset and the VOIP adapter, the message waiting indicator from the VOIP provider seems to no longer get through to the phone. The connection to the VOIP provider is Cable Modem -> Sipura 3002 -> TDM FXO interface -> TDM FXS interface -> phone. Is there a way for Asterisk to get notified and pass the message waiting indicator on to my handset when there is a voice mail waiting at the VOIP provider? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Extension availabilty
I have a client that has an old Merlin system. They would like to move to an Asterisk based system, however, with their existing system each phone is capable of displaying who is on the phone within there office. This is done by lighting a red light for each line(extension) that is in use. Has anyone been able to neatly create this feature? Perhaps an XML application can be written for the Cisco 7960's that would be capable of displaying which extension is being used and which extensions are not in use. Any suggestions would be appreciated. Thanks in advance, -Josh ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sangoma and Digium same machine?
Are you using Digium's new v2 firmware? If not I would recommend against it. I currently have 2 Sangoma quad T1 cards in a single server and it works just fine. Previously I had 2 TE405P(with old firmware) in the machine and had interrupt issues. Replaced with Sangoma boards before Digium v2 firmware was released to fix the problem. Haven't tried 2 Digium quad cards in single system yet. MATT--- On 9/26/05, William Lloyd <[EMAIL PROTECTED]> wrote: Anybody ever put a Sangoma and a Digium card in the same server?Specifically a four port card from each company?-bill[EMAIL PROTECTED]___ --Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk to CCM
Are you using CCM to operate your gateway with MGCP? If so, I had to change the default timers under CCM advanced setup for "Media exchange timers" or the call was timing out at 4 seconds. If the setup was complete prior, it worked fine, but after 4 seconds q.931 from CCM would tear down the call.. On Mon, 2005-09-26 at 14:14 -0300, Arnaldo M. Pereira wrote: > Have you read http://www.voip-info.org/tiki-index.php?page=Asterisk% > 20Cisco%20CallManager%20Integration ? > > I've followed these steps and I can make calls from a CCM client to > Asterisk, but the end point at the Asterisk side can't hear any audio. > > On Mon, 2005-09-26 at 12:28 -0400, Brian J. Rathman wrote: > > I am currently trying to send calls from Asterisk to Cisco Call Manager 4.0 > > and vice versa. I have a SIP trunk setup in CCM and I also have an entry in > > my sip.conf file for CCM. Unfortunately, when I try and send a call to CCM > > I keep getting: > > > > SIP/10.0.0.1-9c18 is circuit-busy > > == Everyone is busy/congested at this time (1:0/1/0) > > -- Got SIP response 481 "Call Leg/Transaction Does Not Exist" back from > > 10.0.0.1 > > > > I have another SIP trunk setup in CCM pointing to my cisco as5300 gateway. > > I can send calls to it and they complete, but when I point the route > > pattern to Asterisk it fails immediatly. Any suggestions? > > > > Thanks, > > Brian > > ___ > > --Bandwidth and Colocation sponsored by Easynews.com -- > > > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SUCCESS - 512 Simultaneous Calls with Digital Recording
Waldo, Thanks for the information. If you don't mind answering: are you guys developing this solution for your internal needs (meaning serving UAs from within your enterprise) or are you planning on offering services to the public? This solution is being developed for our internal needs. It's not that I'm really interested in your business or business model. I'm mainly curious to know how you will deal with potential UAs that are behind external NATs. Will you Asterisk "farm" stand behind a NAT or will it all be "publicly" accessible where no NAT translation or port forwarding will exist? I read the section on Asterisk and NAT on the wiki but still left me with some open questions. Our SIP traffic will never leave our internal network. There will be no NAT/firewalls to traverse. Calls to/from the PSTN will pass through a Cisco AS5400HPX Universal Gateway that handles the TDM/VoIP translation. In my particular setup, I work in a small call center. I have Asterisk behind one NAT with port forwarding on port 5060 and ports 1-2, only because I have 2 remote agents. The rest of the agents are in-house. The remote agents themselves are behind other NATs (behind their DSL service provider). Some times my Asterisk queues have trouble contacting the remote agents. At first, I thought I could simply put a SER server on the public edge, but I'm not sure if that will really solve the problem. I question this setup in terms of stability and security. Even worse, what would happen if my boss decides to increase the remote agents? I spoke to you privately about this and suggested using the IAX protocol with IAXy devices, but you indicated you needed to use SIP. Since we are not dealing with remote agents in our implementation, that is really all I can offer. I hope that the list members will be able to help you solve your problem. Sincerely, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] goiax caller ID
Kevin Scott wrote: I'm not sure what he/she was sending as the caller ID information, what I was trying to do, was send a normal 10 digit number as caller ID. Is there any solution to this? Or anything planned? There are no plans to allow just any caller ID to be sent. Once US dids are available than the DID cid would be sent instead. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Early Media in 180 Ringing
Hello, I have a problem with the following: When I dial a PSTN number from a UAC, the call is made through a SIP Trunk (which has a connection to the PSTN) in Asterisk. The PSTN Gateway returns a 180 Ringing WITH SDP, but Asterisk forwards the 100 Ringing WITHOUT SDP: As you can see below, the SIP message from 10.254.254.1 (the PSTN Gateway) has SDP, while * (with 192.168.0.173) removes the SDP content. How can this be solved? U 10.254.254.1:5060 -> 192.168.0.173:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 192.168.0.173:5060;rport=5060;branch=z9hG4bK454e2d35. Record-Route: . Record-Route: . From: "0161801019" ;tag=as02de1b95. To: ;tag=00-04094-52dbe3bc-6cf68a723. Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE. Contact: . server: Cirpack/v4.38f (gw_sip). Allow: UPDATE, REFER. Content-Type: application/sdp. Content-Length: 253. . v=0. o=cp10 112775383044 112775383045 IN IP4 10.166.38.109. s=SIP Call. c=IN IP4 10.254.254.1. t=0 0. m=audio 35058 RTP/AVP 18 101. b=AS:64. a=rtpmap:18 G729/8000/1. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000/1. a=fmtp:101 0-15. a=ptime:20. # U 192.168.0.173:5060 -> 192.168.1.103:5062 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 192.168.1.103:5062;branch=z9hG4bKff31d98edbf2b265. From: "411" ;tag=f93ee2f65c6906cb. To: ;tag=as675f246d. Call-ID: [EMAIL PROTECTED] CSeq: 60590 INVITE. User-Agent: Asterisk PBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER. Contact: . Content-Length: 0. . ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IptablesAsterisk
I have Asterisk server(1.0.9) behind Iptables firewall. I configured Iptables and sip.conf as below. Andrea(2000) is the outsider phone, on Internet with public IP Luca(2001) is the insider phone, on local network with private IP as well Asterisk server. I noted the ports in play are 5060, 8000, 8001 and 1:2,so to test I put the large rule $IPTABLES -A FORWARD -p udp --dport 8000:2 -j ACCEPT Andrea or Luca receive the rings,but not the voice. Can you help me thank Andrea --- IPTABLES #!/bin/sh IPTABLES=/sbin/iptables # Internal network # LOC_IFACE=eth0 LOC_ADDR=10.100.0.0/24 LOC_IF=10.100.0.1 # External network # EST_IFACE=eth1 EST_ADDR=250.xxx.yyy.24/255.255.255.252 EST_IF=250.xxx.yyy.26 # Asterisk IP and port # PORAST=5060 ASTERISK=10.100.0.225 # deny everything for now # $IPTABLES -P INPUT DROP $IPTABLES -P FORWARD DROP $IPTABLES -P OUTPUT DROP # SIP on UDP port 5060 # $IPTABLES -A FORWARD -i $EST_IFACE -p udp -d $ASTERISK --dport $PORAST -m state --state NEW,ESTABLISHED -j ACCEPT $IPTABLES -A FORWARD -o $EST_IFACE -p udp -s $ASTERISK --sport $PORAST -m state --state ESTABLISHED -j ACCEPT # Other port for phone comunication # $IPTABLES -A FORWARD -p udp --dport 8000:2 -j ACCEPT # Allow from internal to external # $IPTABLES -A FORWARD -o $EST_IFACE -s $LOC_ADDR -m state --state NEW,ESTABLISHED -j ACCEPT $IPTABLES -A FORWARD -i $EST_IFACE -d $LOC_ADDR -m state --state ESTABLISHED -j ACCEPT $IPTABLES -t nat -A POSTROUTING -o $EST_IFACE -j SNAT --to $EST_IF # Asterisk on Internet # $IPTABLES -t nat -A PREROUTING -p udp -d $EST_IF --dport $PORAST -j DNAT --to $ASTERISK:$PORAST --- SIP.CONF [general] port = 5060 bindaddr = 0.0.0.0 allow = all context = bogon-calls [2000] type = friend username = 2000 callerid = Andrea Bencini <2000> secret = 9overthruster7 host = dynamic nat = yes context = from-sip mailbox = 100 [2001] type = friend username = 2001 callerid = Luca Bencini <2001> secret = 11bbanzai9 host = dynamic nat = yes context = from-sip mailbox = 101 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and Broadvoice, no incoming voice
I have been trying on and off for a couple of weeks to no avail... Darren Wright wrote: I am also a long time client, and have no incoming BV today. -Darren http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ? In CLI not working
Has anyone noticed that a ? Entered at the root CLI does not work any longer? Petty I know but I did use it. --john -- This mail was scanned by AntiVir Milter. This product is licensed for non-commercial use. See www.antivir.de for details. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] goiax caller ID
I'm not sure what he/she was sending as the caller ID information, what I was trying to do, was send a normal 10 digit number as caller ID. Is there any solution to this? Or anything planned? Thanks for your time, Kevin ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] I want to send oH323 calls to our Quintum D3000 which is connected to a PSTN
Hi Ade,
An example of oh323.conf is attached, and the lines in the extensions.conf
that make the choice os this oh.323 channel is:
[globals]
GK => OH323/IP of your GK
[local]
ignorepat => 0
exten => _0021NXXX,1,Dial(${GK}/${EXTEN:1})
exten => _0021NXXX,2,Congestion
--
[ ]'s
Daniel Varella de Oliveira
Tecnologia IP Ltda
Tel.: +55 (21)3139-4091 / r. 108
www.tecnologiaip.com.br
On Monday 26 September 2005 13:33, Ade Agbero wrote:
> Hello,
>
>
>
> I want to send oH323 calls to our Quintum D3000.
>
>
>
> I have installed oH323 but I need a working sample oh323.conf and
> extensions.conf, so that I can route specific calls to the Quintum using
> H323.
>
>
>
> For example our Asterisk box IP=192.168.10.100 and Quintum
> IP=192.168.10.101.
>
>
>
> Can anyone assist with a sample Extensions.conf and oH323.conf.
>
>
>
> Thank you,
>
>
>
> Ade.
>
>
>
>
>
>
>
>
> -
> Yahoo! Messenger NEW - crystal clear PC to PC calling worldwide with
> voicemail
;
; Configuration file of OpenH323 channel driver
;
;-
; General configuration options
; (ports, jitter, GK, ...)
;-
[general]
;
; Address to bind to for incoming connections.
; Default is ALL.
;
listenAddress=0.0.0.0
;
; Port to listen to.
; Default value is 1720.
;
listenPort=1720
;
; Configure the TCP port range to be used by H.323
;
tcpStart=1
tcpEnd=2
;
; Configure the UDP port range to be used by H.323
; Note: The port range used by RTP are configured from
; "rtp.conf"
;
udpStart=1
udpEnd=2
;
; Enable fast start (yes,no).
;
fastStart=no
;
; Enable H.245 tunnelling (yes,no).
;
h245Tunnelling=no
;
; Enable early H.245 messages in call SETUP message.
;
h245inSetup=no
;
; Enable in-band-DTMF detection.
; (Note: Netmeeting uses in-band DTMFs)
;
inBandDTMF=no
;
; Set jitter buffer (in milliseconds, 20...1).
;
jitterMin=20
jitterMax=100
;
; Set IP Type-of-Service byte for RTP channels.
; Valid values for this option are:
; lowdelay, throughput, reliability, mincost, none
; Moreover, an integer (in decimal or hex format) may be entered.
;
ipTos=none
;
; Set the maximum number of inbound/outbound/simultaneous
; H.323 connections.
;
outboundMax=10
inboundMax=10
simultaneousMax=10
;
; Call Rate Limiter params (ingress direction). When the total number
; of active calls is above 'crlThreshold' then the rate of the incoming
; H.323 calls is restricted in a way where no more than 'crlCallNumber'
; calls are allowed in 'crlCallTime' milliseconds, thus limiting the rate
; of incoming calls to:
; 'crlCallNumber' / ('crlCallTime' / 1000) Calls-per-Sec.
;
;crlCallNumber=20
;crlCallTime=2
;crlThreshold=30
;
; Set the bandwidth limit for H.323 connections.
; The value is in Kbps.
;
;bandwidthLimit=1024
;
; Set tracing options for the wrapper library and for the
; OpenH323 library.
; libTraceFile can be 'stdout' or a full path name to the tracefile.
; Only the trace info for OpenH323 is logged in libTraceFile.
;
wrapLibTraceLevel=3
libTraceLevel=3
;libTraceFile=stdout
libTraceFile=/var/log/asterisk-h323.log
;
; Disable gatekeeper or specify a gatekeeper. The gatekeeper's ID is the zone
name.
; Valid values for this option are:
; DISABLE,
; DISCOVER,
; ,
; ,
; GKID:
; @
;
gatekeeper=IP of your gatekeeper
;gatekeeper=DISABLE
;
; Set the gatekeeper password. If used, it enables H.235 access to gatekeeper.
;
;gatekeeperPassword=secret
;
; Set the gatekeeper registration timeout. Before the expiration of
; the timeout, a re-registration is attempted.
;
gatekeeperTTL=600
;
; Set the mode for sending user-input (DTMF)
; Valid values for this option are:
; Q931- Q.931 Keypad Information Element
; STRING - H.245 string
; TONE- H.245 tone
; RFC2833 - RFC2833
;
userInputMode=TONE
;
; AMA flags (default, omit, billing, documentation)
;
amaFlags=default
;
; Account code
;
accountCode=H323
;
; Default language
;
language=en
;
; Default Music-On-Hold class
;
musiconhold=default
;
; Set the default context of H.323 calls.
;
context=voip-h323
;-
; Configure H.323 aliases, prefixes and
; related ASTERISK's contexts
;-
[register]
;
; Aliases/prefixes associated with the default context
; defined in section [general].
;
;alias=asterisk
;alias=123
;
; Aliases/prefixes routed in "all-aliases" context.
;
context=all-aliases
alias=asterisk
alias=99001701
alias=99001702
;
;
; Aliases/prefixes routed in "more-aliases" context.
;
;context=more-aliases
;alias=665
;
; Aliases/prefixes routed in "all-prefixes" context.
;
;context=all-prefixes
;gwprefix=00
;gwprefix=01
;
; Aliases/prefixes routed in "more-stuff" context.
;
;context=more-stuff
;alias=664
;gwprefix=02
;[cisco2]
;type=h323
;e164=02124950937
;context=all-aliases
;-
; Specify and configure CODEC related
; op
Re: [Asterisk-Users] change codec based on callerid (sip/iax)
This can be done by modifying the source code. trixter http://www.0xdecafbad.com wrote: I have been asked if asterisk can change codecs dynamically based on the calling party's caller id. I couldnt find anything, and dont know that this is something that asterisk can do, but it occurs to me that possibly with a reinvite it can be done, however I dont think you can issue those from the dialplan or agi. The only solution I can think of on this is to use something like ser (www.iptel.org/ser) in between the asterisk box and forward effectivly to a different account on the asterisk box based on caller id (ie ser makes a choice which account to use). codecs then would be negotiated normally at connect time. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FW: channel offhook state
Yes indeed. There have been huge changes to chan_zap.c in CVS-HEAD compared to 1.09. In 1.09 Stable there are a lot of problems with handling call hang-ups. CVS-HEAD, of 28/08 was much better. But even though it did improve things, it wasn't quite right. In particular I found two problems with polarity reversal detection in chan_zap.c for which I have created a patch (this is now in CVS-HEAD). Please see http://bugs.digium.com/view.php?id=5191 for more details. Please note that you'll need to use answeronpolarityswitch=yes and/or hanguponpolarityswitch=yes in your Zapata.conf to make full use of the polarity detection code. You will also need to be very careful if CID is sent on a polarity switch too -- you may need to make it detect on the 0th ring or you could suffer from immediate hang-ups on ring. Unfortunately I've received a problem report with this modification. Any updates Magnus? I'm hoping it is all down to the ring that CID is detected at, and that by changing it to 0 or 1 all will be well again. But anybody who has had problems with hangup detection in the past should try CVS-HEAD and play with the options above to see if it improves things. Having said all this, things are still not perfect: For UK (and possibly other European countries) we still require a way for Asterisk to detect the continuous tone that indicates a remote party hangup on a POTS line. The Sipura 3000 uses this method and I believe it works quite well, though I've not tried it myself. Faris. -Original Message- FWIW, there were a couple of channel zap changes made in the last couple of days to cvs-head. Don't have a clue whether those fixes addressed the problem you're talking about. > Has anyone else experienced the same problem, where a Zap channel gets stuck > in off-hook state? > > Thanks > > > -Original Message- > > From: [EMAIL PROTECTED] > > [mailto:[EMAIL PROTECTED] > > Sent: Friday, September 23, 2005 1:45 PM > > To: asterisk-users@lists.digium.com > > Subject:[Asterisk-Users] FW: channel offhook state > > > > > > > > -Original Message- > > From: Jacqueline Lee [mailto:[EMAIL PROTECTED] > > Sent: Friday, September 23, 2005 11:46 AM > > To: asterisk-users@lists.digium.com > > Subject:channel offhook state > > > > > > We are using a digium card (TDM400) with asterisk for our access to the > > PSTN. Initially when the server starts, all the zap channels on the card > > are in the "onhook" state. As soon as a channel is used (for inbound or > > outbound PSTN calls) the corresponding channel goes into "offhook" state, > > and stays in "offhook" state, even after the call ends; Asterisk log shows > > that the channel was hungup. Most of the time, the channel is still usable > > to make more PSTN calls, even though it shows in "offhook" state. > > Occasionally the channel becomes unusable for making PSTN calls (usually > > channel 1). The symptom is Asterisk and the client show the PSTN call was > > established, but the destination PSTN number never really receives the > > call. > > > > Shouldn't the channel go back to "onhook" state once the call hangs up? Is > > the persistent "offhook" state causing the channel to eventually become > > unusable? > > ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk and Broadvoice, no incoming voice
I am also a long time client, and have no incoming BV today.
-Darren
From: [EMAIL PROTECTED] on behalf of Jason Schafer
Sent: Mon 9/26/2005 12:08 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk and Broadvoice, no incoming voice
> Does asterisk says something in the verbose console?
I'm not sure what the verbose console is, but I can run sip debug and
post the output when I make an inbound call.
> please post your sip.conf relevant entries for BroadVoice.
[general]
port = 5060 ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0; Address to bind to (all addresses on machine)
disallow=all
allow=ulaw
allow=alaw
context = from-sip-external ; Send unknown SIP callers to this context
callerid = Unknown
pedantic=no ; added for Broadvoice support 8/3/05 EK
externip=216.xxx.xxx.xxx
localnet=172.xxx.xxx.0/255.255.255.0
I have just
> cancelled with BroadVoice (too much latency for the places i wanted to
> call), so i never used the incoming number. But im glad to help if i can.
I have outbound setup on VOIPJet, my intent with the Broadvoice is to
setup a forward on busy with my landline to roll over to the BV number.
Here's the output from sip debug
m=audio 14008 RTP/AVP 0 8 2 18 96 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:18 G729/8000
a=rtpmap:96 iLBC/8000
a=rtpmap:101 telephone-event/8000
13 headers, 12 lines
Using latest request as basis request
Sending to 147.135.0.128 : 5060 (non-NAT)
Found no matching peer or user for '147.135.0.128:5060'
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 2
Found RTP audio format 18
Found RTP audio format 96
Found RTP audio format 101
Peer audio RTP is at port 147.135.0.128:14008
Found description format PCMU
Found description format PCMA
Found description format G726-32
Found description format G729
Found description format iLBC
Found description format telephone-event
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x51c
(ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined -
0x1 (g723)
Looking for s in from-sip-external
list_route: hop:
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 147.135.0.128:5060;branch=z9hG4bK3qd0e4102041k9sak0k1.1sr
From: "Schafer Trish
";tag=SD28clb01-1612693231-1127750324179
To: "Jason Schafer"
Call-ID: SD28clb01-d697b8d4cd8742e341c1f1942d1bf7e1-js11002
CSeq: 160704490 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact:
Content-Length: 0
to 147.135.0.128:5060
-- Executing Wait("SIP/147.135.0.129-095da350", "1") in new stack
-- Executing Goto("SIP/147.135.0.129-095da350", "from-pstn|s|1") in
new stack
-- Goto (from-pstn,s,1)
-- Executing GotoIf("SIP/147.135.0.129-095da350",
"1?from-pstn-reghours|s|1:") in new stack
-- Goto (from-pstn-reghours,s,1)
-- Executing GotoIf("SIP/147.135.0.129-095da350",
"0?from-pstn-reghours-nofax|s|1:2") in new stack
-- Goto (from-pstn-reghours,s,2)
-- Executing Answer("SIP/147.135.0.129-095da350", "") in new stack
We're at 216.xxx.xxx.xxx port x
Answering with preferred capability 0x4 (ulaw)
Answering with preferred capability 0x8 (alaw)
Answering with non-codec capability 0x1 (telephone-event)
Reliably Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 147.135.0.128:5060;branch=z9hG4bK3qd0e4102041k9sak0k1.1sr
From: "Schafer Trish
";tag=SD28clb01-1612693231-1127750324179
To: "Jason
Schafer";tag=as2a994d31
Call-ID: SD28clb01-d697b8d4cd8742e341c1f1942d1bf7e1-js11002
CSeq: 160704490 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact:
Content-Type: application/sdp
Content-Length: 238
v=0
o=root 1782 1782 IN IP4 216.xxx.xxx.xxx
s=session
c=IN IP4 216.xxx.xxx.xxx
t=0 0
m=audio 14138 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
to 147.135.0.128:5060
-- Executing Wait("SIP/147.135.0.129-095da350", "1") in new stack
asterisk1*CLI>
Sip read:
ACK sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 147.135.0.128:5060;branch=z9hG4bK3qd0u4103gtgb94c0080.1sr
From: "Schafer Trish
";tag=SD28clb01-1612693231-1127750324179
To: "Jason
Schafer";tag=as2a994d31
Call-ID: SD28clb01-d697b8d4cd8742e341c1f1942d1bf7e1-js11002
CSeq: 160704490 ACK
Contact:
Max-Forwards: 69
Content-Length: 0
9 headers, 0 lines
-- Executing SetVar("SIP/147.135.0.129-095da350", "intype=aa_2") in
new stack
-- Executing Cut("SIP/147.135.0.129-095da350", "intype=intype|-|1")
in new stack
-- Executing GotoIf("SIP/147.135.0.129-095da350", "0?7:9") in new stack
-- Goto (from-pstn-reghours,s,9)
-- Executing GotoIf("SIP/147.135.0.129-095da350", "0?10:12") in new
stack
-- Goto (from-pstn-reghours,s,12)
-- Executing GotoIf("SIP/147.135.0.129-095da350"
Re: [Asterisk-Users] Tux/Asterisk logo for Cisco phones
El jue, 22-09-2005 a las 19:04, David McNett escribió: > I made http://slacker.com/~nugget/stuff/asterisk-cow-real.bmp for my > non-Linux asterisk servers. I made my * + tux + office logo http://www.cipher.com.pe/central/asterisk-tux-cipher.bmp Regards, -- Ing CIP Alejandro Celi Mariátegui <[EMAIL PROTECTED]> ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk to CCM
Have you read http://www.voip-info.org/tiki-index.php?page=Asterisk% 20Cisco%20CallManager%20Integration ? I've followed these steps and I can make calls from a CCM client to Asterisk, but the end point at the Asterisk side can't hear any audio. On Mon, 2005-09-26 at 12:28 -0400, Brian J. Rathman wrote: > I am currently trying to send calls from Asterisk to Cisco Call Manager 4.0 > and vice versa. I have a SIP trunk setup in CCM and I also have an entry in > my sip.conf file for CCM. Unfortunately, when I try and send a call to CCM I > keep getting: > > SIP/10.0.0.1-9c18 is circuit-busy > == Everyone is busy/congested at this time (1:0/1/0) > -- Got SIP response 481 "Call Leg/Transaction Does Not Exist" back from > 10.0.0.1 > > I have another SIP trunk setup in CCM pointing to my cisco as5300 gateway. I > can send calls to it and they complete, but when I point the route pattern to > Asterisk it fails immediatly. Any suggestions? > > Thanks, > Brian > ___ > --Bandwidth and Colocation sponsored by Easynews.com -- > > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- Arnaldo M. Pereira egghunt at gmail dot com http://ansi-c.org/~arnaldo ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
