Re: [Asterisk-Users] show channel issue with 1.2.9
On Tue, 2006-06-06 at 11:37 +0600, [EMAIL PROTECTED] wrote: try asterisk -rx 'show channels' that is what I did try, yes I ommited the quotes in the email guess it wasnt understood that it returns only the header and not any information on what channels are in use nor any information on how many active or total calls are in progress. This works upto about 50 channels, after that it starts to break. This worked fine with over 200 channels with 1.2.4 however its very unreliable with 1.2.9. So something was changed. If I do just asterisk -r then type show channels it appears to always work, its just when its done from the shell prompt that it doesnt. It acts almost like a race condition that 'wins' when the channel count is low, but looses almost always when it gets to a moderate level. Why I was thinking it was a threading issue. -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.trxtel.com we pay you to terminate calls with us! signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] More Level QueueSystem
Hi Patrick, Let me see if I am following you here. When a caller calls in, obviously you want them to be in the first queue level based on your dial plan. Now, how do you want the caller to reach the next queue? Is the only way a caller going to go to the next queue via a transfer from the level 1 attendant? If so, I would make the dial plan like this: 123,1,Answer() 123,2,Queue(1stLevel,t) 124,1,Answer() 124,2,Queue(2ndLevel,t) 125,1,Answer() 125,2,Queue(3rdLevel,t) This provides a few different things that it looks like you are going for. One, it will allow separation of each queue level. So when the attendant in level 1 needs to transfer to a level 2, they just transfer to the new extension and the caller is moved to the new queue. Also, if say queue 1 is closed, this will prevent callers from gaining access to higher queue levels. Also you can add NoOP statments to record items, or an AGI script as well before the caller enters the queue so you know what happened. The return codes are as follows: 0 means that the queue is full, emtpy (no members present) or doesn't exist. -1 means that caller hung upbut if the call is bridged then it means either of the parties could have stopped the call. 1 I think means the caller entered the queue without a problem. I don't think that will be returned. At least that is how I understood everything. Kevin Patrick Bök wrote: Hi, I am trying to set up a dial plan und I have a few problems to realise some functions. The dial plan should look like this: 123,1,Answer() 123,2,Queue(1stlevel,t) 123,3,Queue(2ndlevel,t) 123,4,Queue(3rdlevel,t) 123,5,Hangup() If a member of the 1stlevel-Queue can answer the call it should be hanged up after finishing. If not, the current member answering the call should be able to transfer the caller to the 2ndlevel-Queue. And so on. How can I check whether it is transfered or hanged up? I do not know how to realise this workflow, the transfer, within the dial plan and I have not found any solution within the Wiki. The next problem I have got with the queue app is the value of the return code: 0 for not being answered -1 for hangup 1 for bridged (does bridge in this context mean the same as transfer???) Would be nice if you could help me about the transfer problem between the queues. Thanks a lot, Patrick ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Config Revision Control
On 12:33, Mon 05 Jun 06, Douglas Garstang wrote: I guess this is wy beyond my knowledge of subversion. I just started playing with the directory structure I might use, and first thought was something like this: [EMAIL PROTECTED] ~/cfg $ ls -l total 16 drwxr-xr-x 2 dougg users 4096 Jun 5 12:24 acd drwxr-xr-x 2 dougg users 4096 Jun 5 12:28 common drwxr-xr-x 2 dougg users 4096 Jun 5 12:28 pbx drwxr-xr-x 2 dougg users 4096 Jun 5 12:24 vm where acd, pbx and vm refer to a function, or class of systems. pbx/ would have systems pbx1, pbx2 and pbx3 beneath it. Some files, such as sound files, and AGI are common to all systems, and hence the common/ directory. However, I have no idea what to do with it beyond that. I don't know how to push common changes out to all the other servers, or inherit, or whatever, or how to stop a common directory being created on the servers instead of putting the files from common under /var/lib/asterisk/agi-bin and /usr/lib/asterisk/sounds etc. Arrgh. To push the common changes you need to setup the automerge script. To checkout multiple trees inside one you can use the svn properties. There's this property svn:external. You can read more about it in the svnbook. Good luck. -- Michiel van Baak http://michiel.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Why is it drug addicts and computer afficionados are both called users? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Local vs. toll Dial Plan
At 10:58 PM 6/5/2006, you wrote: exten = _310.,1,NoOp It can get tricky if several pattern cover the same range. But I odn't believe that this is the case. There's likely lots of ways to do what he wants, I thought he asked for a solution to a particular problem, matching a 3 digit number against a list of 3 digit numbers in a way that's reasonably easy to maintain. Other than the list is a bit of trouble to maintain but easy to generate automatically, my solution, while a bit convoluted is short, simple and reasonably east to maintain. If it helps him I'm happy, if not it made me think a bit and that's always worthwhile. It came out of an attempt to make an automatic list processing engine to find the best possible outgoing line in my setup. You'll find that under GROUP() on the wiki and any ideas you might have to improve would be gladly accepted. Ira ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] How many TE405 ...
Hi, It is impossible for CPU to do so heavy task. Is there anybodyuse4E1 card work well on one PC server? For the PCI limited and the software dsp, the voice quanlity will be not acceptable. Could anyone give me some evidence? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of ArdSent: Tuesday, June 06, 2006 5:30 AMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] How many TE405 ... Hi, Is it possible to use 4 TE405 boards in one server ? It mean, to have 16 E1s on just one server. Can somebody tell me how many boards is itpossible to have on one server ? Thanks, ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ISDN BRI (I.430) over ethernet
I also want to know this. This would be very useful in Call center for remote attendent. The E1 gateway will do this very well, but where is the BRI voip gateway? Hawk -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of James Harper Sent: Tuesday, June 06, 2006 10:13 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] ISDN BRI (I.430) over ethernet Does anyone know of a hardware adapter that can take ISDN BRI frames (I.430) and encapsulate them in Ethernet (any form, but TDMoE would be really cool), in much the same way that the redfone does for PRI? (yes I have asked this before in looser terms, but it was a while ago :) Would anyone find such a device useful? Thanks James ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] STNU spport
Hi,all. There have any STUN spport for asterisk? thanks,,,-- Jeffery `∧ ∧︵ ミ^r^ミ灬)~iaxtel Num: 1-700-576-1311fwdnet Num: 728150http://www.diaip.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] STNU spport
On Tue, 2006-06-06 at 15:17 +0800, Chen Fan wrote: Hi,all. There have any STUN spport for asterisk? thanks,,, where asterisk queries a stun server or where asterisk acts like a stun server? Becuase stun is totally self contained it would be silly (in my opinion anyway) to have a stun server built in. There are many free ones out there, stunner is one example. As for stun client support, afaik asterisk doesnt really do that yet. and it should have a periodic timeout if it does, but it would be a chan_sip addition. There is also an RTP patch that gets rid on 99% of NAT problems with SIP by technically violating the RFC but given the way networks work 99.99% of the time it would work perfectly (so if it were an option it would be good for everyone). What it does is on the RTP port if it receives a packet it will use that IP instead of whatever is specified. This fixes NAT problems on the other end at least, but for some reason it never made it into the source tree :/ -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.trxtel.com we pay you to terminate calls with us! signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_capi-cm-0.6 and incoming calls problem
James and Armin, Turn on asterisk debugging too. Capi seems to be working okay, maybe asterisk isn't picking up the call for some reason. Maybe: asterisk -r set verbose 9 set debug 9 capi debug then make an incoming call and copy the output into an email and send it to the list (unless it is really really long, then you may have to look for interesting bits). u should see a message in there somewhere that tells you that either the capi driver is rejecting the call because it doesn't want to answer that msn (your earlier logs make that unlikelye), or that asterisk can't find an extension for it. James Thanks for your response I managed to get it working with the following as you suggested, [capi-in] exten = 99546476,1,Dial(Sip/123,20) exten = 99546476,2,Voicemail(123) exten = 99546476,3,Hangup Following is the debug output when it started working, -- ISDN1: info element CHANNEL IDENTIFICATION 89 INFO_IND ID=001 #0x09d5 LEN=0016 Controller/PLCI/NCCI= 0x101 InfoNumber = 0xa1 InfoElement = a1 INFO_RESP ID=001 #0x09d5 LEN=0012 Controller/PLCI/NCCI= 0x101 -- ISDN1: info element Sending Complete -- ISDN1: CAPI/ISDN1/99546476-16: 99546476 matches in context capi-in -- Executing Dial(CAPI/ISDN1/99546476-16, Sip/123|20) in new stack == Started pbx on channel CAPI/ISDN1/99546476-16 -- Called 123 -- SIP/123-0b4d is ringing == ISDN1: Requested RINGING-Indication for CAPI/ISDN1/99546476-16 I now have two other problems, 1. Noise is quite loud using this line with asterisk 2. Incoming DTMF tones are not detected via this ISDN line and AVM Fritz card Any ideas on how to overcome these issues? Esteban _ Research and compare new cars side by side at carpoint.com.au http://secure-au.imrworldwide.com/cgi-bin/a/ci_450304/et_2/cg_801459/pi_1004813/ai_833884 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Query: IAXModem
Hi, I am in a problem. Can anybody help me out. I am trying to establish connection using hyperterminal through IAXsoft modem using asterisk PBX. I have done the following settings in the configuraion files of asterisk. 1) iax.conf file: [iaxmodem] type=friend ;username=iaxmodem ;secret=n19d19 host=dynamic qualify=yes ;trunk=yes ;context=in-fax disallow=all allow=ulaw allow=alaw allow=gsm [iaxmodem1] type=friend ;type=peer ;username=iaxmodem ;secret=n19d19 host=dynamic qualify=yes ;trunk=yes ;context=in-fax ;not required allow=ulaw allow=alaw allow=gsm 2)extensions.conf file exten = 12,1,Dial(IAX2/iaxmodem1) //for dialing to modem2 with extension number 12 through modem1 using hyperterminal 3) created ttyIAX0 and ttyIAX1 file in path /etc/iaxmodem ttyIAX0 file: device /dev/ttyIAX0 owner uucp:uucp mode 660 port 4571 refresh 100 server 127.0.0.1 peername iaxmodem secret password #Cidname John0 #Cidnumber 8005551212 codec slinear ttyIAX1 file: device /dev/ttyIAX1 owner sanchal:uucp mode 660 port 4572 refresh 300 server 127.0.0.1 peername iaxmodem1 #secret password cidname John2 cidnumber 8005551231 codec slinear 4)Now using hyper terminal send atdt12 from one side it sends ring to other side . On replying ATA from other side, it sends connect but not in accordance with class1 format. Client other end at+fclass=1 -- -- at+fclass=1 OK ---- OK atdt12 -- -- ring connect -- -- ATA -- connect NOCARRIER -- --ERROR Can anybody give me the guidelines how to proceed further to transfer a file after establishing a successful connection. Regards sanchal ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ISDN BRI (I.430) over ethernet
Try to see this... http://www.patapsco.co.uk/applications/isdn_conversion_sharing_and_simulation/share_pris/share_pri.htm On 6/6/06, Hao Xu [EMAIL PROTECTED] wrote: I also want to know this. This would be very useful in Call center for remote attendent. The E1 gateway will do this very well, but where is the BRI voip gateway?Hawk -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of James Harper Sent: Tuesday, June 06, 2006 10:13 AM To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] ISDN BRI (I.430) over ethernetDoes anyone know of a hardware adapter that can take ISDN BRI frames(I.430) and encapsulate them in Ethernet (any form, but TDMoE would be really cool), in much the same way that the redfone does for PRI? (yes I have asked this before in looser terms, but it was a while ago :)Would anyone find such a device useful?ThanksJames___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Chanspy Jitter?
Wes Baehr ha scritto: (Sometimes) When I’m monitoring calls, I hear a very bad jitter – usually only on one of the bridged channels. So at first I thought it was just the one end of the conversation actually causing the jitter – but it’s not. So I called in from another device to spy at the same time – and the other chanspy sounds perfectly normal. (And neither party is complaining of bad sound) So, periodically, chanspy seems to lose sync with its source – has anyone else had this problem? Running 1.2.7.1, calls are all SIP-IAX2 Me, same problem, same version, all possible combinations of SIP/IAX calls. Looking at iax2 netstats shows there's no real problem, jitter, delay, packet loss are well within acceptable limits, still what I hear is echoed robo-voice ! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Help - DTMF feedthru
I thought I had this fix but I just upgraded from 1.2.7 to 1.2.9 and it seems to be happenning again Using Sipura SPA-3000 with *. I want DTMF tones to go thru when I call out on PSTN - I.E. if I call my bank or external VM and need to put out DTMF. I found with 1.2.7 if I turned off all transfer and other local DTMF options - did not put any flags in dial it would work fine. Now in 1.2.9 it seems to have broken again. I have not changed anything (no flags) and the DTMF is not getting thru. Any ideas? Doug * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307* * * * [EMAIL PROTECTED]* * http://www.crompton.com * ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] STNU spport
hi, We need STUN client support for asterisk... becasue the service provider only offer STUN interface,, so i can not connect asterisk to their server i have found that there someone is develop res_stun.c ..but still not release... regards On 6/6/06, trixter aka Bret McDanel [EMAIL PROTECTED] wrote: On Tue, 2006-06-06 at 15:17 +0800, Chen Fan wrote: Hi,all. There have any STUN spport for asterisk? thanks,,,where asterisk queries a stun server or where asterisk acts like a stunserver?Becuase stun is totally self contained it would be silly (in my opinionanyway) to have a stun server built in.There are many free ones out there, stunner is one example.As for stun client support, afaik asterisk doesnt really do that yet.and it should have a periodic timeout if it does, but it would be achan_sip addition.There is also an RTP patch that gets rid on 99% of NAT problems with SIP by technically violating the RFC but given the way networks work 99.99%of the time it would work perfectly (so if it were an option it would begood for everyone).What it does is on the RTP port if it receives a packet it will use that IP instead of whatever is specified.This fixesNAT problems on the other end at least, but for some reason it nevermade it into the source tree :/--Trixter http://www.0xdecafbad.com Bret McDanelBelfast IE +44 28 9099 6461DE +49 801 777 555 3402Utrecht NL +31 306 553058US WA +1 360 207 0479US NY +1 516 687 5200FreeWorldDialup: 635378 http://www.trxtel.com we pay you to terminate calls with us!-BEGIN PGP SIGNATURE-Version: GnuPG v1.4.3 (GNU/Linux)iD8DBQBEhTCp+1olxlzQw5cRAlMcAJ9kRP+x6pEF3FlQj1KQj+vXJNx7XwCfUAw+ Rh9enc6pLooaEai9EgLC5jQ==StPS-END PGP SIGNATURE-___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users -- Jeffery `∧ ∧︵ ミ^r^ミ灬)~iaxtel Num: 1-700-576-1311fwdnet Num: 728150http://www.diaip.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] STNU spport
On Tue, 2006-06-06 at 16:12 +0800, Chen Fan wrote: hi, We need STUN client support for asterisk... becasue the service provider only offer STUN interface,, so i can not connect asterisk to their server all stun does is resolve your external IP by sending data to a foreign server which looks at the IP and returns it back to you. It has nothing to do with the channel used other than SIP will then use that IP (which can be defined by either externhost or externip - dont forget localnet too in sip.conf). i have found that there someone is develop res_stun.c ..but still not release... likely that is just going to replace the externip value in the chan_sip driver. I cant imagine that it would do much more than that. Have you set both externip and localnet in sip.conf and checked to see if that works? If you dont do NAT on your end it wont even be required. -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.trxtel.com we pay you to terminate calls with us! signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] STNU spport
HI, There is a parameter NAT can be set in the configuration file. Is it the way that we can use to support NAT by setting nat=yes in the file instead using other NAT resolving tools like stun? On 6/6/06, trixter aka Bret McDanel [EMAIL PROTECTED] wrote: On Tue, 2006-06-06 at 16:12 +0800, Chen Fan wrote: hi, We need STUN client support for asterisk... becasue the service provider only offer STUN interface,, so i can not connect asterisk to their server all stun does is resolve your external IP by sending data to a foreign server which looks at the IP and returns it back to you. It has nothing to do with the channel used other than SIP will then use that IP (which can be defined by either externhost or externip - dont forget localnet too in sip.conf). i have found that there someone is develop res_stun.c ..but still not release... likely that is just going to replace the externip value in the chan_sip driver. I cant imagine that it would do much more than that. Have you set both externip and localnet in sip.conf and checked to see if that works? If you dont do NAT on your end it wont even be required. -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.trxtel.com we pay you to terminate calls with us! -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.3 (GNU/Linux) iD8DBQBEhUFA+1olxlzQw5cRAoDtAKCK5ufDIpmsXG/p2ydcj3VDqxA7jgCcCAHi bpFsVQ8FJuxF+crAEm2hwZE= =VQtX -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Can I use an onboard modem?
Just a quick question: Is there a driver for a normal modem to be used as an FXS line (to connect a normal analogue phone to your PC)? Thanks, Pieter ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can I use an onboard modem?
Which chip set? Cheers, Madhawa pieter Claassen wrote: Just a quick question: Is there a driver for a normal modem to be used as an FXS line (to connect a normal analogue phone to your PC)? Thanks, Pieter ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can I use an onboard modem?
On Tue, Jun 06, 2006 at 11:09:28AM +0200, pieter Claassen wrote: Just a quick question: Is there a driver for a normal modem to be used as an FXS line (to connect a normal analogue phone to your PC)? A normal modem may serve as an FXO line with the proper hardware, as it is basically a phone. Also chances are that the on-board modem has no such driver for Asterisk. But if you show its entry from lspci, it may help. -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 [EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dialstatus
Hi, I use an E1-Board to hand the calls over to internal SIP-Clients. My Question is which Dialstatus is set when the SIP-client is unreachable. I tried with NOANSWER but does not seem to be suitable. Does anyone of you have a solution? In voip-info.org wiki there is a Dialstatus CHANUNAVAIL but this is explained by Channel unavailable. On SIP, peer may not be registered.. So this seems not to be right, or does it? TIA, Christophorus begin:vcard fn:Christophorus Laube n:Laube;Christophorus org:SemanticEdge GmbH adr:;;Kaiserin-Augusta-Allee 10-11;Berlin;;10553;Deutschland email;internet:[EMAIL PROTECTED] title:Systemadministrator tel;work:+49-30-34507758 url:http://www.semanticedge.de version:2.1 end:vcard ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: 2 rings after making a phone call
Hi All, I have setup [EMAIL PROTECTED] 2.8 and using Digium TDM400P cards Whenever I dial out and finish the conversation and put the SIP Snom320 phone down, it rings back twice? If you pick up the phone there is no answer.although you think it's a genuine call!! I have attached the logs; please can anyone help on how to stop this. Regards Ash == Spawn extension (macro-dialout-trunk, s, 14) exited non-zero on 'SIP/200-c98a' in macro 'dialout-trunk' == Spawn extension (macro-dialout-trunk, s, 14) exited non-zero on 'SIP/200-c98a' -- Executing Macro(SIP/200-d6c5, dialout-trunk|1|90775x||) in new stack -- Executing GotoIf(SIP/200-d6c5, 1?3:2)) in new stack -- Goto (macro-dialout-trunk,s,3) -- Executing Macro(SIP/200-d6c5, user-callerid) in new stack -- Executing DBget(SIP/200-d6c5, AMPUSER=DEVICE/200/user) in new stack -- DBget: varname=AMPUSER, family=DEVICE, key=200/user -- DBget: set variable AMPUSER to 200 -- Executing DBget(SIP/200-d6c5, AMPUSERCIDNAME=AMPUSER/200/cidname) in new stack -- DBget: varname=AMPUSERCIDNAME, family=AMPUSER, key=200/cidname -- DBget: set variable AMPUSERCIDNAME to Reception -- Executing GotoIf(SIP/200-d6c5, 0?5) in new stack -- Executing SetCallerID(SIP/200-d6c5, Reception 200) in new stack -- Executing NoOp(SIP/200-d6c5, Using CallerID Reception 200) in new stack -- Executing Macro(SIP/200-d6c5, record-enable|200|OUT) in new stack -- Executing GotoIf(SIP/200-d6c5, 0 0?2:4) in new stack -- Goto (macro-record-enable,s,4) -- Executing AGI(SIP/200-d6c5, recordingcheck|20060606-110927|1149588567.614) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck recordingcheck|20060606-110927|1149588567.614: Outbound recording not enabled -- AGI Script recordingcheck completed, returning 0 -- Executing NoOp(SIP/200-d6c5, No recording needed) in new stack -- Executing Macro(SIP/200-d6c5, outbound-callerid|1) in new stack -- Executing DBget(SIP/200-d6c5, USEROUTCID=AMPUSER/200/outboundcid) in new stack -- DBget: varname=USEROUTCID, family=AMPUSER, key=200/outboundcid -- DBget: set variable USEROUTCID to -- Executing GotoIf(SIP/200-d6c5, 0?4) in new stack -- Executing SetCallerID(SIP/200-d6c5, 02077292040) in new stack -- Executing GotoIf(SIP/200-d6c5, 1?6) in new stack -- Goto (macro-outbound-callerid,s,6) -- Executing NoOp(SIP/200-d6c5, CallerID set to 02077292040) in new stack -- Executing SetGroup(SIP/200-d6c5, OUT_1) in new stack -- Executing CheckGroup(SIP/200-d6c5, ) in new stack -- Executing SetVar(SIP/200-d6c5, DIAL_NUMBER=90775x) in new stack -- Executing SetVar(SIP/200-d6c5, DIAL_TRUNK=1) in new stack -- Executing AGI(SIP/200-d6c5, fixlocalprefix) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix fixlocalprefix: Removed prefix. New number: 0775xx -- AGI Script fixlocalprefix completed, returning 0 -- Executing SetVar(SIP/200-d6c5, OUTNUM=0775xxx) in new stack -- Executing Cut(SIP/200-d6c5, custom=OUT_1|:|1) in new stack -- Executing GotoIf(SIP/200-d6c5, 0?16) in new stack -- Executing Dial(SIP/200-d6c5, ZAP/g0/0775xxx) in new stack -- Called g0/0775 -- Zap/1-1 answered SIP/200-d6c5 -- Hungup 'Zap/1-1' == Spawn extension (macro-dialout-trunk, s, 14) exited non-zero on 'SIP/200-d6c5' in macro 'dialout-trunk' == Spawn extension (macro-dialout-trunk, s, 14) exited non-zero on 'SIP/200-d6c5' == Starting post polarity CID detection on channel 1 -- Starting simple switch on 'Zap/1-1' -- Executing Set(Zap/1-1, FROM_DID=s) in new stack -- Executing Goto(Zap/1-1, ext-group|1|1) in new stack -- Goto (ext-group,1,1) -- Executing Macro(Zap/1-1, user-callerid|) in new stack -- Executing DBget(Zap/1-1, AMPUSER=DEVICE//user) in new stack -- DBget: varname=AMPUSER, family=DEVICE, key=/user -- DBget: Value not found in database. -- Executing DBget(Zap/1-1, AMPUSERCIDNAME=AMPUSER//cidname) in new stack -- DBget: varname=AMPUSERCIDNAME, family=AMPUSER, key=/cidname -- DBget: Value not found in database. -- Executing GotoIf(Zap/1-1, 1?5) in new stack -- Goto (macro-user-callerid,s,5) -- Executing NoOp(Zap/1-1, Using CallerID ) in new stack -- Executing GotoIf(Zap/1-1, 0?NEWPREFIX) in new stack -- Executing Set(Zap/1-1, CALLERID(name)=) in new stack -- Executing Set(Zap/1-1, RGPREFIX=) in new stack -- Executing Set(Zap/1-1, CALLERID(name)=) in new stack -- Executing Set(Zap/1-1, RecordMethod=Group) in new stack -- Executing Macro(Zap/1-1, record-enable||Group) in new stack -- Executing GotoIf(Zap/1-1, 0 0?2:4) in new stack -- Goto (macro-record-enable,s,4) -- Executing AGI(Zap/1-1, recordingcheck|20060606-110944|1149588584.616) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck recordingcheck
[Asterisk-Users] Playback welcome message while phones ring, please help
I want that incoming callers to hear a welcome message while the phones ring. I know I can use Dial with the m(class) option to make the same with musiconhold, but the problem is that musiconhold does not start from the beginning of my mp3 file. If I use Playback or Background, the phones do not ring unless the mp3 file is over... Any suggestion? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ISDN BRI (I.430) over ethernet
PRI is primary rate ISDN, consisting of (normally) 30 channels or (rarely - US only I think) 24 channels. BRI is basic rate ISDN, and consists of 2 channels. They are not the same thing. The redfone is a PRI to TDMoE converter, I'm after something that does the same thing for BRI. Thanks James -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Shenen Shenen Sent: Tuesday, 6 June 2006 17:55 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] ISDN BRI (I.430) over ethernet Try to see this... http://www.patapsco.co.uk/applications/isdn_conversion_sharing_and_simul at ion/share_pris/share_pri.htm On 6/6/06, Hao Xu [EMAIL PROTECTED] wrote: I also want to know this. This would be very useful in Call center for remote attendent. The E1 gateway will do this very well, but where is the BRI voip gateway? Hawk -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk- [EMAIL PROTECTED] On Behalf Of James Harper Sent: Tuesday, June 06, 2006 10:13 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] ISDN BRI (I.430) over ethernet Does anyone know of a hardware adapter that can take ISDN BRI frames (I.430) and encapsulate them in Ethernet (any form, but TDMoE would be really cool), in much the same way that the redfone does for PRI? (yes I have asked this before in looser terms, but it was a while ago :) Would anyone find such a device useful? Thanks James ___ --Bandwidth and Colocation provided by Easynews.com http://easynews.com/ -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com http://easynews.com/ -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Change in dial command behaviour between 1.2.7.1 and 1.2.8?
Hi list! Are there any changes in the behaviour of the Dial command between 1.2.7.1 and 1.2.8.? I am forwarding calls to my legacy PBX using : exten = s,1,Dial(Zap/g1/8210,90,r) Ever since I upgraded to 1.2.9 it seems as if the Legacy PBX is no longer receiving the extension I am calling on the PBX and the call gets dropped to the switchboard extension on the legacy PBX. Did I goof up or did something change? Thanks!! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Realtime and SIP Registration
Hi!I use the following configuration to register my asterisk server to my SIP provider:register = 12345:[EMAIL PROTECTED]/12345sip.conf :[sipout-test]type=peerusername=12345fromuser=12345fromdomain=provider.comsecret=passwdinsecure=veryhost=sip.provider.com qualify=yescontext=test-incomingextensions.conf:exten = 12345,1,Dial(SIP/10)exten = _0NXZXX,1,Dial(SIP/[EMAIL PROTECTED])This works fine when I put it into the config files. I can dial other numbers via my provider and receive calls. Wenn I put everything into Realtime tables (except the register command), incoming calls work only after * I make at least one outgoing call - or - * Somebody calls me twiceOn incoming calls, the caller first gets a 'user unavailale' from my SIP provider. When hanging up and calling again, the connection establishes successfully and I see this when entering 'sip show peers': sipout-test/12345 IP.AD.DR.ESS 5060 UNKNOWNThis line does not show up when I registering my phone to my asterisk server. But it shows up immediately after registerung the phone when I use config files instead of RTA. I don't know wheter this is RTA- or a config-problem. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Playback welcome message while phones ring, please help
I believe if you use the new native music on hold feature it always plays the music on hold starting from the beginning. On Tue, 2006-06-06 at 11:15, Tommaso Calosi wrote: I want that incoming callers to hear a welcome message while the phones ring. I know I can use Dial with the m(class) option to make the same with musiconhold, but the problem is that musiconhold does not start from the beginning of my mp3 file. If I use Playback or Background, the phones do not ring unless the mp3 file is over... Any suggestion? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Local vs. toll Dial Plan
I am working on an AGI script to do just this. The idea is to use the XML database search at localcallingguide.com and decide if the call is local or LD based on parameters that you provide to the script. Will keep you posted on the progress of this. I'm hoping to have it done soon. The problem has been finding the time to work on it. -Jason Doug Crompton wrote: Ok asked this earlier with no response so I will phrase it a different way. I am sure someone had to deal with this and there is a best way. I want to let Asterisk make the decision on best path based on local exchange - xxx-yyy - where xxx is one of my local area codes and xxx is exchange designator. The problem is that the list is rather large. Maybe 50-100. The idea is that I can talor my local PSTN usage to only those exchanges that are non-toll for me. Other calls will go out ENUM to best way. So is there a way to do this without entering a dialplan for every exchange designator? Doug * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307* * * * [EMAIL PROTECTED]* * http://www.crompton.com * ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can´t send emails
Tzafir:This was the result of the test you gave me. Obviously the message was not sent.On my environment I have a mail server and the [EMAIL PROTECTED] server. I just want to take one email out of the box, no matter if it is to internet or inside my network. Thanks for your help and please keep going.What do you think would happen if both the mail server and the [EMAIL PROTECTED] have the same name, would be a cause of this kind of problems? Thanks. View Message Message Number: 271 Inbox | Compose | Reply |Reply All | Forward | Delete | Logout Date: Tuesday,June 06, 200607:27 AM From: Mail Delivery Subsystem [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Returned mail: see transcript for details The original message was received at Tue, 6 Jun 2006 07:27:39 -0400 from [EMAIL PROTECTED] - The following addresses had permanent fatal errors - [EMAIL PROTECTED] (reason: 550 5.1.1 [EMAIL PROTECTED]... User unknown) (expanded from: [EMAIL PROTECTED]) - Transcript of session follows - ... while talking to [127.0.0.1]: DATA 550 5.1.1 [EMAIL PROTECTED]... User unknown 550 5.1.1 [EMAIL PROTECTED] User unknown 503 5.0.0 Need RCPT (recipient) message/delivery-status: Type Type:unknown Date: Tuesday,June 06, 200607:27 AM From: root root To: [EMAIL PROTECTED] Subject: hola uknown: Type Type:unknown Tzafrir Cohen [EMAIL PROTECTED] escribió: On Mon, Jun 05, 2006 at 01:45:08PM -0500, yrving rivas wrote: Lewis: This is what the logs says regarding to de mails on a test I made: Jun 5 14:28:59 DEBUG[27498] app_voicemail.c: Sent mail to [EMAIL PROTECTED] with command '/usr/sbin/sendmail -t' Later will come with an error like this: Date: Monday, June 05, 2006 12:54 PM From: Mail Delivery Subsystem <[EMAIL PROTECTED]> To: [EMAIL PROTECTED] Subject: Postmaster notify: see transcript for details The original message was received at Wed, 31 May 2006 12:32:34 -0400 from localhost with id k4VGSP31004351 - The following addresses had permanent fatal errors - <[EMAIL PROTECTED]> This is probably wrong.Should the mail be sent to an external server or to a server in yourLAN?Can you send mails using a simple:echo test | mail -s "test subject" [EMAIL PROTECTED]-- Tzafrir Cohen sip:[EMAIL PROTECTED]icq#16849755 iax:[EMAIL PROTECTED]+972-50-7952406 [EMAIL PROTECTED] http://www.xorcom.com___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __Correo Yahoo!Espacio para todos tus mensajes, antivirus y antispam ¡gratis! Regístrate ya - http://correo.yahoo.com.mx/ __Correo Yahoo!Espacio para todos tus mensajes, antivirus y antispam ¡gratis! Regístrate ya - http://correo.yahoo.com.mx/ ___ --Bandwidth and Colocation provided by Easynews.com --
[Asterisk-Users] What to do on a national celebration day? Test, test, test!
Today is Swedens national day - since a few years a holiday too. We don't have a tradition on how to celebrate. Sweden has not been to war for a very long time, so there's no real spirit for the country here - it's been aroundfor such a long time, so what? :-) Guess we have to learn from abroad, to get a celebration feeling like July 4th in the US or May 17th in Norway (from reading the recipe I got from my friend Eivind, I am a bit hesitant about it really something we want to do)... I'll try to fill the day somehow. We've raised the flag outside. We have friends coming over. The grill is ready. The beer too... Meanwhile, I have to get you folks back on track, back to testing. Don't try to run away now, we need you! I'll relief you from test-this-branch for now. The differences between that branch and trunk is not that big any more, and what remains either won't make it into 1.4 or will be integrated this week. I will soon start to add more stuff to test-this-branch, continue to make it a scary place to be. Instead of testing the test branch, please test svn trunk. We will soon produce a 1.4 beta based on Asterisk svn trunk. Download instructions are to be found on http://www.asterisk.org We have integrated quite a lot of big changes during the last week for you to test and turn inside out. If you locate bugs, please report them in the bug tracker. - The followme application - a toolkit for creating findme/followme extensions - RTCP support improvements - necessary for video calls - A generic jitterbuffer for zap, sip, skinny, h.323, iax2 and other channels - Totally rewritten SIP transfer code In testing right now is the new t38 passthrough code, which will be integrated after some more reviews and tests. Please help us test that code too. Thank you all for working hard on this until I return online! Regards, /Olle --- * Olle E. Johansson - [EMAIL PROTECTED] * Asterisk Training http://edvina.net/training/ * Next training and dCAP in Stockholm, Sweden, June 2006! --- * Olle E Johansson - [EMAIL PROTECTED] * Cell phone +46 70 593 68 51, Office +46 8 96 40 20, Sweden ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] PABX Setup
Hi, We are trying to port over a PABX to our network. Both PRI's seem to be live however, whenever someone dials out from the PABX Asterisk happens to report : -- Extension '' in context 'samsungincoming' from '736327438' does not exist. Rejecting call on channel 0/31, span 2 If crc4 is turned off, it reports a yellow alarm. Any suggestions? Regards, Sahil Gupta VoiceValley ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can´t send emails
I will look out for that software. I´ll let you know. Thank you.YrvingAlex Robar [EMAIL PROTECTED] escribió: Yrving,OK, so you've definetly got a sendmail configuration problem. If you're not big on config files, try using Webmin. It's a web-based interface for thousands of administrative tasks for Linux systems (including configuring webmin). It lays out all the options for you, and tells you what each of them means. You should be able to get a working system based on that. AlexOn 6/5/06, yrving rivas [EMAIL PROTECTED] wrote: Nope. Nothing out of the box.thanks for your help..thanks!YrvingAlex Robar [EMAIL PROTECTED] escribió: Yrving,You can send to a local user on the system, but can you send to an external account? AlexOn 6/5/06, yrving rivas [EMAIL PROTECTED] wrote:Alex:I verified my SMTP with "telnet ( [EMAIL PROTECTED] ip) 25" and sent an email to root. I have another mail server inside my network and, as happens trying to send an email to internet, it gives me a time out.I can give you all the details if you would. Thanks for your help.yrvingAlex Robar [EMAIL PROTECTED] escribió: Yrving,How have you verified that your SMTP is working? Can you actually send a message from your Asterisk server to one of your internal addresses? Alex On 6/5/06, yrving rivas [EMAIL PROTECTED] wrote: Hello everybody.I will apreciate your help in this case.I have the environment showed in the picture of the file included in this message.I would like to send trhough internet all the messages received by voicemail. I made all the configuration through the AMP, and all messages are saved properly, so I can receive them directly from my phone with *97, but still the Asterisk don´t send it through Internet. As I have seen the SMTP is working, and I can ping any site of Internet, and any of my internal addresses, but still I have no idea about what is happening.Thanks.Yrving__Correo Yahoo!Espacio para todos tus mensajes, antivirus y antispam ·gratis! Regístrate ya -http://correo.yahoo.com.mx/___--Bandwidth and Colocation provided by Easynews.com--Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Alex Robar [EMAIL PROTECTED] ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __Correo Yahoo!Espacio para todos tus mensajes, antivirus y antispam ·gratis! Regístrate ya - http://correo.yahoo.com.mx/ ___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Robar[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users __Correo Yahoo!Espacio para todos tus mensajes, antivirus y antispam ¡gratis! Regístrate ya - http://correo.yahoo.com.mx/ ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Alex Robar[EMAIL PROTECTED] ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __Correo Yahoo!Espacio para todos tus mensajes, antivirus y antispam ¡gratis! Regístrate ya - http://correo.yahoo.com.mx/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] PABX Setup
Samsung PABX? Its TEPRI probably configured in overlap mode so you need to configure asterisk span that is connected to PABX to overlap mode as well. When user selects the outside line in overlap mode PABX connects to asterisk and then sends the digits to it as the user presses the key's. If overlap mode is not configured in asterisk switch is not started by asterisk and it just thinks that empty dial string was sent to it. Just use: overlapdial=yes in your zapata.conf Make sure you have exten = s,1,Busy() exten = s,2,Hangup in your 'samsungincoming' context so that users get a busy signal when they didn't enter any digits in allotted time otherwise you'll get a hanging channel in Samsung. We use that setup with OfficeServ 500 and it works really well. Regards -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sahil Gupta Sent: Tuesday, 6 June 2006 21:40 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] PABX Setup Hi, We are trying to port over a PABX to our network. Both PRI's seem to be live however, whenever someone dials out from the PABX Asterisk happens to report : -- Extension '' in context 'samsungincoming' from '736327438' does not exist. Rejecting call on channel 0/31, span 2 If crc4 is turned off, it reports a yellow alarm. Any suggestions? Regards, Sahil Gupta VoiceValley ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] PABX Setup
Thanks mate. All going well. Regards, Sahil Gupta VoiceValley On Tue, 6 Jun 2006, Boris Bakchiev wrote: Samsung PABX? Its TEPRI probably configured in overlap mode so you need to configure asterisk span that is connected to PABX to overlap mode as well. When user selects the outside line in overlap mode PABX connects to asterisk and then sends the digits to it as the user presses the key's. If overlap mode is not configured in asterisk switch is not started by asterisk and it just thinks that empty dial string was sent to it. Just use: overlapdial=yes in your zapata.conf Make sure you have exten = s,1,Busy() exten = s,2,Hangup in your 'samsungincoming' context so that users get a busy signal when they didn't enter any digits in allotted time otherwise you'll get a hanging channel in Samsung. We use that setup with OfficeServ 500 and it works really well. Regards -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sahil Gupta Sent: Tuesday, 6 June 2006 21:40 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] PABX Setup Hi, We are trying to port over a PABX to our network. Both PRI's seem to be live however, whenever someone dials out from the PABX Asterisk happens to report : -- Extension '' in context 'samsungincoming' from '736327438' does not exist. Rejecting call on channel 0/31, span 2 If crc4 is turned off, it reports a yellow alarm. Any suggestions? Regards, Sahil Gupta VoiceValley ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Personal Inquiry
Does any of these asterisk users know, one Mr Jeffry from Kochi, India, Please let me know Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] weather
Please any one knows how to configure the weather on asterisk or if there a weather channel I can subscribe to it * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] weather
[EMAIL PROTECTED]/Trixbox includes built-in weather AGI scripts. You should have some good luck getting weather on your Asterisk box if you look into those scripts.AlexOn 6/6/06, Khaled Chehab [EMAIL PROTECTED] wrote: Please any one knows how to configure the weather on asterisk or if there a weather channel I can subscribe to it * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Alex Robar[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] weather
http://nerdvittles.com/index.php?p=134On 6/6/06, Khaled Chehab [EMAIL PROTECTED] wrote: Please any one knows how to configure the weather on asterisk or if there a weather channel I can subscribe to it * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ISDN BRI (I.430) over ethernet
CPE-Asterisk(NT Mode)-ip-Asterisk(CPE)-NT? maybe it will fit for you? if yes, i think you can work with the following budget: via epia board ~85$ mini itx case (small size!) ~85$ ram ~20$ DiskOnChip (or HDD) ~20 - ~50 HFC BRI ~50$ so globally ~300-350/side you can also go for patton something of ~800$ BRI is basic rate ISDN, and consists of 2 channels. They are not the same thing. The redfone is a PRI to TDMoE converter, I'm after something that does the same thing for BRI. Does anyone know of a hardware adapter that can take ISDN BRI frames (I.430) and encapsulate them in Ethernet (any form, but TDMoE would be really cool), in much the same way that the redfone does for PRI? (yes I have asked this before in looser terms, but it was a while ago :) -- WoodOO-[P]an[G]alaktikan[A]gent-People ][ http://shadow.pganet.com [EMAIL PROTECTED]@RedHat.users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Compile install error.
- Doug Crompton [EMAIL PROTECTED] wrote: I am getting the following error at the end of 'make install' 1.2.9 I have not tried to find it but I suspect there is just a misplaced punctuation. It runs fine. That part of the Makefile has not been touched in quite a while, so there should not be any behavioral differences from 1.2.8, 1.2.7.1, etc. If this is repeatable on your system, please open a bug report on bugs.digium.com. -- Kevin P. Fleming Senior Software Engineer Digium, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] show channel issue with 1.2.9
- trixter aka Bret McDanel [EMAIL PROTECTED] wrote: It acts almost like a race condition that 'wins' when the channel count is low, but looses almost always when it gets to a moderate level. Why I was thinking it was a threading issue. I believe this has been a known problem for a while, where 'asterisk -rx' does not reliably wait until the output has been generated/flushed before exiting the remote console process. Why it would have suddenly started affecting you I can't guess :-) -- Kevin P. Fleming Senior Software Engineer Digium, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How many TE405 ...
On Tuesday 06 June 2006 02:56, Hao Xu wrote: It is impossible for CPU to do so heavy task. Is there anybody use 4E1 card work well on one PC server? For the PCI limited and the software dsp, the voice quanlity will be not acceptable. Could anyone give me some evidence? YATE has multiple systems running four quad Sangoma E1 cards (16 E1s) without issue. Obviously I think the transcoding is kept to a minimum, but it is possible. -A. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Change in dial command behaviour between 1.2.7.1 and 1.2.8?
- Remco Barendse [EMAIL PROTECTED] wrote: Did I goof up or did something change? No, there should not be any behavioral changes between 1.2.7.1 and 1.2.8 except for bug fixes. The only change that I see in the ChangeLog for 1.2.8 that could be relevant is the one regarding 'usecallingpres' handling chan_zap; are you using that option in your zapata.conf file? -- Kevin P. Fleming Senior Software Engineer Digium, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_capi-cm-0.6 and incoming calls problem
On Tue, 6 Jun 2006, Esteban Guana-Jarrin wrote: James and Armin, Turn on asterisk debugging too. Capi seems to be working okay, maybe asterisk isn't picking up the call for some reason. Maybe: asterisk -r set verbose 9 set debug 9 capi debug then make an incoming call and copy the output into an email and send it to the list (unless it is really really long, then you may have to look for interesting bits). u should see a message in there somewhere that tells you that either the capi driver is rejecting the call because it doesn't want to answer that msn (your earlier logs make that unlikelye), or that asterisk can't find an extension for it. James Thanks for your response I managed to get it working with the following as you suggested, [capi-in] exten = 99546476,1,Dial(Sip/123,20) exten = 99546476,2,Voicemail(123) exten = 99546476,3,Hangup Following is the debug output when it started working, -- ISDN1: info element CHANNEL IDENTIFICATION 89 INFO_IND ID=001 #0x09d5 LEN=0016 Controller/PLCI/NCCI= 0x101 InfoNumber = 0xa1 InfoElement = a1 INFO_RESP ID=001 #0x09d5 LEN=0012 Controller/PLCI/NCCI= 0x101 -- ISDN1: info element Sending Complete -- ISDN1: CAPI/ISDN1/99546476-16: 99546476 matches in context capi-in -- Executing Dial(CAPI/ISDN1/99546476-16, Sip/123|20) in new stack == Started pbx on channel CAPI/ISDN1/99546476-16 -- Called 123 -- SIP/123-0b4d is ringing == ISDN1: Requested RINGING-Indication for CAPI/ISDN1/99546476-16 Looks good. I now have two other problems, 1. Noise is quite loud using this line with asterisk Maybe you should adapt the gains in capi.conf. 2. Incoming DTMF tones are not detected via this ISDN line and AVM Fritz card Did you set for softdtmf/relaxdtmf? Armin ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dialstatus
Wether the SIP client is not registered or does not exists at all you will get CHANUNAVAIL. Regards On 6/6/06, Christophorus Laube [EMAIL PROTECTED] wrote: Hi, I use an E1-Board to hand the calls over to internal SIP-Clients. My Question is which Dialstatus is set when the SIP-client is unreachable. I tried with NOANSWER but does not seem to be suitable. Does anyone of you have a solution? In voip-info.org wiki there is a Dialstatus CHANUNAVAIL but this is explained by Channel unavailable. On SIP, peer may not be registered.. So this seems not to be right, or does it? TIA, Christophorus ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Change in dial command behaviour between 1.2.7.1 and 1.2.8?
On Tue, 6 Jun 2006, Kevin P. Fleming wrote: - Remco Barendse [EMAIL PROTECTED] wrote: Did I goof up or did something change? No, there should not be any behavioral changes between 1.2.7.1 and 1.2.8 except for bug fixes. The only change that I see in the ChangeLog for 1.2.8 that could be relevant is the one regarding 'usecallingpres' handling chan_zap; are you using that option in your zapata.conf file? Yes I do have usecallingpres=yes in my zaptel.conf. I already noticed this and tried commenting out this line and did a reload on * but that didn't change much. I'll try downgrading zaptel first, if it doesn't work I will also downgrade asterisk back to 1.2.7.1 Thanks! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dialstatus
I tried with CHANUNAVAIL but I was not successful. I want to try to call a SIP client. If it is not answering and cannot be found I want wo call someone else. How can I do that? NOANSWER and CHANUNAVAIL do not work out. Wether the SIP client is not registered or does not exists at all you will get CHANUNAVAIL. Regards On 6/6/06, Christophorus Laube [EMAIL PROTECTED] wrote: Hi, I use an E1-Board to hand the calls over to internal SIP-Clients. My Question is which Dialstatus is set when the SIP-client is unreachable. I tried with NOANSWER but does not seem to be suitable. Does anyone of you have a solution? In voip-info.org wiki there is a Dialstatus CHANUNAVAIL but this is explained by Channel unavailable. On SIP, peer may not be registered.. So this seems not to be right, or does it? TIA, Christophorus ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Prices of g729 codec
- Steve Underwood [EMAIL PROTECTED] wrote: Asterisk should really import a recent version of Speex. The last time I checked it had an ancient version. Quality has improved, and computation has significantly reduced. As far as I know, Asterisk has never included the Speex library in its source package, it has always used the one present on the system at build time. GSM and iLBC yes, Speex no (although the SVN trunk version now prefers the system's GSM library if one is present). -- Kevin P. Fleming Senior Software Engineer Digium, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dialstatus
this is what I have, and it works on Asterisk-1.2.1
[macro-sipextens]
exten = s,1,Macro(validate_extension)
exten = s,2,Dial(SIP/${sipprefix}${num},${calltimeout}|${calloptions})
exten = s,3,Macro(catch_dial_response,${DIALSTATUS})
so, After Dial, I catch the dial response, and heres the catch macro
[macro-catch_dial_response]
exten = s,1,GotoIf($[${ARG1} = NOANSWER]? 11 : 2)
exten = s,2,GotoIf($[${ARG1} = CHANUNAVAIL] ? 22 : 3)
exten = s,3,GotoIf($[${ARG1} = BUSY]? 33 : 4)
exten = s,4,Macro(generic_handler)
exten = s,11,Macro(noanswer_handler)
exten = s,22,Macro(unavail_handler)
exten = s,33,Macro(busy_handler)
FInally here are the 4 other macros
[macro-noanswer_handler]
exten = s,1,SetCDRUserField(-10/${agi_cdr_id})
exten = s,2,Set(voicemail_flags=u)
exten = s,3,Playback(iss_noanswer_channel_${defaultlang})
exten = s,4,Goto(loopback_ivr,s,1)
[macro-unavail_handler]
exten = s,1,SetCDRUserField(-11/${agi_cdr_id})
exten = s,2,Set(voicemail_flags=u)
exten = s,3,GotoIf($[foo${is_exten} = foo] ? 4 : 6)
exten = s,4,Playback(iss_unavailable_channel_${defaultlang})
exten = s,5,Goto(loopback_ivr,s,1)
exten = s,6,Playback(iss_unavailable_extension_${defaultlang})
exten = s,7,Goto(loopback_ivr,s,1)
[macro-busy_handler]
exten = s,1,SetCDRUserField(-12/${agi_cdr_id})
exten = s,2,Set(voicemail_flags=b)
exten = s,3,Playback(iss_busy_channel_${defaultlang})
exten = s,4,Goto(loopback_ivr,s,1)
[macro-generic_handler]
exten = s,1,SetCDRUserField(-14/${agi_cdr_id})
exten = s,2,Set(voicemail_flags=u)
exten = s,3,GotoIf($[foo${is_exten} = foo] ? 4 : 6)
exten = s,4,Playback(iss_unavailable_channel_${defaultlang})
exten = s,5,Goto(loopback_ivr,s,1)
exten = s,6,Playback(iss_unavailable_extension_${defaultlang})
exten = s,7,Goto(loopback_ivr,s,1)
If you cant get it working, simply do something like this:
[test]
exten = _XX,1,Answer()
exten = _XX,2,Dial(SIP/${EXTEN})
exten = _XX,3,NoOp(${DIALSTATUS})
That will tell you what status is generated.
Regards
On 6/6/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
I tried with CHANUNAVAIL but I was not successful. I want to try to call a
SIP client. If it is not answering and cannot be found I want wo call
someone else.
How can I do that? NOANSWER and CHANUNAVAIL do not work out.
Wether the SIP client is not registered or does not exists at all you
will get CHANUNAVAIL.
Regards
On 6/6/06, Christophorus Laube [EMAIL PROTECTED] wrote:
Hi,
I use an E1-Board to hand the calls over to internal SIP-Clients. My
Question is which Dialstatus is set when the SIP-client is unreachable.
I tried with NOANSWER but does not seem to be suitable.
Does anyone of you have a solution?
In voip-info.org wiki there is a Dialstatus CHANUNAVAIL but this is
explained by Channel unavailable. On SIP, peer may not be
registered.. So this seems not to be right, or does it?
TIA, Christophorus
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Re: [Asterisk-Users] Change in dial command behaviour between 1.2.7.1 and 1.2.8?
On 15:46, Tue 06 Jun 06, Remco Barendse wrote: On Tue, 6 Jun 2006, Kevin P. Fleming wrote: - Remco Barendse [EMAIL PROTECTED] wrote: Did I goof up or did something change? No, there should not be any behavioral changes between 1.2.7.1 and 1.2.8 except for bug fixes. The only change that I see in the ChangeLog for 1.2.8 that could be relevant is the one regarding 'usecallingpres' handling chan_zap; are you using that option in your zapata.conf file? Yes I do have usecallingpres=yes in my zaptel.conf. I already noticed this and tried commenting out this line and did a reload on * but that didn't change much. I'll try downgrading zaptel first, if it doesn't work I will also downgrade asterisk back to 1.2.7.1 Try a restart first. A reload may not be enough. -- Michiel van Baak http://michiel.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Why is it drug addicts and computer afficionados are both called users? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FW: voice mail
I am using [EMAIL PROTECTED] v 2.6 I want to active or deactivate voicemail from command line Like database put AMPUSER/VOICEMAIL/111 ENABLE this command is applicable at version 2.8 but it don't work at 2.6 Any one can help me ?? Regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dialstatus
Check out this example dialplan: http://pastebin.ca/19456
That should give you everything you need.
bp
On 6/6/06, Moises Silva [EMAIL PROTECTED] wrote:
this is what I have, and it works on Asterisk-1.2.1[macro-sipextens]exten = s,1,Macro(validate_extension)
exten = s,2,Dial(SIP/${sipprefix}${num},${calltimeout}|${calloptions})exten = s,3,Macro(catch_dial_response,${DIALSTATUS})so, After Dial, I catch the dial response, and heres the catch macro
[macro-catch_dial_response]exten = s,1,GotoIf($[${ARG1} = NOANSWER]? 11 : 2)exten = s,2,GotoIf($[${ARG1} = CHANUNAVAIL] ? 22 : 3)exten = s,3,GotoIf($[${ARG1} = BUSY]? 33 : 4)exten = s,4,Macro(generic_handler)
exten = s,11,Macro(noanswer_handler)exten = s,22,Macro(unavail_handler)exten = s,33,Macro(busy_handler)FInally here are the 4 other macros[macro-noanswer_handler]exten = s,1,SetCDRUserField(-10/${agi_cdr_id})
exten = s,2,Set(voicemail_flags=u)exten = s,3,Playback(iss_noanswer_channel_${defaultlang})exten = s,4,Goto(loopback_ivr,s,1)[macro-unavail_handler]exten = s,1,SetCDRUserField(-11/${agi_cdr_id})
exten = s,2,Set(voicemail_flags=u)exten = s,3,GotoIf($[foo${is_exten} = foo] ? 4 : 6)exten = s,4,Playback(iss_unavailable_channel_${defaultlang})exten = s,5,Goto(loopback_ivr,s,1)exten = s,6,Playback(iss_unavailable_extension_${defaultlang})
exten = s,7,Goto(loopback_ivr,s,1)[macro-busy_handler]exten = s,1,SetCDRUserField(-12/${agi_cdr_id})exten = s,2,Set(voicemail_flags=b)exten = s,3,Playback(iss_busy_channel_${defaultlang})
exten = s,4,Goto(loopback_ivr,s,1)[macro-generic_handler]exten = s,1,SetCDRUserField(-14/${agi_cdr_id})exten = s,2,Set(voicemail_flags=u)exten = s,3,GotoIf($[foo${is_exten} = foo] ? 4 : 6)
exten = s,4,Playback(iss_unavailable_channel_${defaultlang})exten = s,5,Goto(loopback_ivr,s,1)exten = s,6,Playback(iss_unavailable_extension_${defaultlang})exten = s,7,Goto(loopback_ivr,s,1)
If you cant get it working, simply do something like this:[test]exten = _XX,1,Answer()exten = _XX,2,Dial(SIP/${EXTEN})exten = _XX,3,NoOp(${DIALSTATUS})That will tell you what status is generated.
RegardsOn 6/6/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: I tried with CHANUNAVAIL but I was not successful. I want to try to call a
SIP client. If it is not answering and cannot be found I want wo call someone else. How can I do that? NOANSWER and CHANUNAVAIL do not work out. Wether the SIP client is not registered or does not exists at all you
will get CHANUNAVAIL. Regards On 6/6/06, Christophorus Laube [EMAIL PROTECTED] wrote: Hi,
I use an E1-Board to hand the calls over to internal SIP-Clients. My Question is which Dialstatus is set when the SIP-client is unreachable. I tried with NOANSWER but does not seem to be suitable.
Does anyone of you have a solution? In voip-info.org wiki there is a Dialstatus CHANUNAVAIL but this is explained by Channel unavailable. On SIP, peer may not be
registered.. So this seems not to be right, or does it? TIA, Christophorus ___
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Re: [Asterisk-Users] Local vs. toll Dial Plan
OK Great. I will wait for your success! Are you using Perl? Doug On Tue, 6 Jun 2006, Jason Bachman wrote: I am working on an AGI script to do just this. The idea is to use the XML database search at localcallingguide.com and decide if the call is local or LD based on parameters that you provide to the script. Will keep you posted on the progress of this. I'm hoping to have it done soon. The problem has been finding the time to work on it. -Jason Doug Crompton wrote: Ok asked this earlier with no response so I will phrase it a different way. I am sure someone had to deal with this and there is a best way. I want to let Asterisk make the decision on best path based on local exchange - xxx-yyy - where xxx is one of my local area codes and xxx is exchange designator. The problem is that the list is rather large. Maybe 50-100. The idea is that I can talor my local PSTN usage to only those exchanges that are non-toll for me. Other calls will go out ENUM to best way. So is there a way to do this without entering a dialplan for every exchange designator? Doug * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307* * * * [EMAIL PROTECTED]* * http://www.crompton.com * ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Those that sacrifice essential liberty to obtain a little temporary safety deserve neither liberty nor safety. -- Ben Franklin (1759) * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307* * * * [EMAIL PROTECTED]* * http://www.crompton.com * ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: *** Spam *** [Asterisk-Users] Example config files for Snom mass updating?
Hi, maybe this helps ? http://www.snom.com/wiki/index.php/DHCP http://www.snom.com/wiki/index.php/Massdeployment_Firmware_Release_5 For further questions in that regard feel free to contact us at [EMAIL PROTECTED] ! Regards. On Friday 02 June 2006 05:49, Remco Barendse wrote: Hi list! Does anyone have a small tarball with some example config files for Snom mass updating under linux? The Snom docs are not very clear to me and they nicely specify some DHCP parameter numbers(??) and how to do that when running windoze servers but the linux info is very thin. Undoubtedly there will be a lot of people that already have this working with dhcpd and the various types of Snom phones, could anyone share their dhcpd and Snom specific update files? Thanks!! -- --- See our Docs, FAQs, etc at: http://snom.com/wiki --- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Change in dial command behaviour between 1.2.7.1 and 1.2.8?
On Tue, 6 Jun 2006, Michiel van Baak wrote: On 15:46, Tue 06 Jun 06, Remco Barendse wrote: On Tue, 6 Jun 2006, Kevin P. Fleming wrote: - Remco Barendse [EMAIL PROTECTED] wrote: Did I goof up or did something change? No, there should not be any behavioral changes between 1.2.7.1 and 1.2.8 except for bug fixes. The only change that I see in the ChangeLog for 1.2.8 that could be relevant is the one regarding 'usecallingpres' handling chan_zap; are you using that option in your zapata.conf file? Yes I do have usecallingpres=yes in my zaptel.conf. I already noticed this and tried commenting out this line and did a reload on * but that didn't change much. I'll try downgrading zaptel first, if it doesn't work I will also downgrade asterisk back to 1.2.7.1 Try a restart first. A reload may not be enough. Thanks, I'll also try that but it's kind of difficult on a production box :) But nevertheless, I couldn't find any info (or at least info that makes sense to me) :) about `usecallingpres' but setting it to yes would be the right setting in my case? I just want it to show everything to anybody :) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Compiling VD_app_conference for x86_64
Hi Erick. Just for record I could compile the application but could not
use. It was crashing the asterisk.
Then I downloaded the 0.5 release and used the 64_Makefile to compile.
Now its working on test environment but until now everything is ok.
(http://optusnet.dl.sourceforge.net/sourceforge/astguiclient/VD_app_conference_0.5.zip)
Thanks for your help and save the list!
Rgds,
Ricardo.
Erick Perez wrote:
They key point is to disable de x86 CFLAGS and add this one
CFLAGS += -march=k8 -fPIC
k8 is the machine type for x86_64
On 6/4/06, Erick Perez [EMAIL PROTECTED] wrote:
This is my makefile, it compiled ok. I will test it tomorrow but if
you have somewhere to test today, let me know.
# $Id: Makefile,v 1.9 2005/10/27 17:53:35 stevek Exp $
#
# Makefile, based on the Asterisk Makefile, Coypright (C) 1999, Mark
Spencer
#
# Copyright (C) 2002,2003 Junghanns.NET GmbH
#
# Klaus-Peter Junghanns [EMAIL PROTECTED]
#
# This program is free software and may be modified and
# distributed under the terms of the GNU Public License.
#
.EXPORT_ALL_VARIABLES:
#
# app_conference defines which can be passed on the command-line
#
INSTALL_PREFIX := /usr
INSTALL_MODULES_DIR := $(INSTALL_PREFIX)/lib/asterisk/modules
ASTERISK_INCLUDE_DIR :=
$(HOME)/sources/asterisk02/asterisk-1.2.8/include
# turn app_conference debugging on or off ( 0 == OFF, 1 == ON )
APP_CONFERENCE_DEBUG := 1
# 0 = OFF 1 = astdsp 2 = speex
SILDET := 2
#
# app_conference objects to build
#
OBJS = app_conference.o conference.o member.o frame.o cli.o
SHAREDOS = app_conference.so
#
# standard compile settings
#
PROC = $(shell uname -m)
INSTALL = install
CC = gcc
INCLUDE = -I$(ASTERISK_INCLUDE_DIR)
LIBS = -ldl -lpthread -lm
DEBUG := -g
CFLAGS = -pipe -Wall -Wmissing-prototypes -Wmissing-declarations
$(DEBUG) $(INCLUDE) -D_REENTRANT -D_GNU_SOURCE
#CFLAGS += -O2
#CFLAGS += -O3 -march=pentium3 -msse -mfpmath=sse,387 -ffast-math
# PERF: below is 10% faster than -O2 or -O3 alone.
#CFLAGS += -O3 -ffast-math -funroll-loops
# below is another 5% faster or so.
CFLAGS += -O3 -ffast-math -funroll-all-loops -fprefetch-loop-arrays
-fsingle-precision-constant
# this is fun for PPC
#CFLAGS += -mcpu=7450 -faltivec -mabi=altivec -mdynamic-no-pic
# this is fun for x86
# The line below was commented by Erick Perez [EMAIL PROTECTED]
#CFLAGS += -march=pentium3 -msse -mfpmath=sse,387
# adding -msse -mfpmath=sse has little effect.
#CFLAGS += -O3 -msse -mfpmath=sse
#CFLAGS += $(shell if $(CC) -march=$(PROC) -S -o /dev/null -xc
/dev/null /dev/null 21; then echo -march=$(PROC); fi)
CFLAGS += $(shell if uname -m | grep -q ppc; then echo
-fsigned-char; fi)
CFLAGS += -DCRYPTO
# The line below was added by Erick Perez [EMAIL PROTECTED]
CFLAGS += -march=k8 -fPIC
ifeq ($(APP_CONFERENCE_DEBUG), 1)
CFLAGS += -DAPP_CONFERENCE_DEBUG
endif
#
# additional flag values for silence detection
#
ifeq ($(SILDET), 2)
OBJS += libspeex/preprocess.o libspeex/misc.o libspeex/smallft.o
CFLAGS += -Ilibspeex -DSILDET=2
endif
ifeq ($(SILDET), 1)
CFLAGS += -DSILDET=1
endif
OSARCH=$(shell uname -s)
ifeq (${OSARCH},Darwin)
SOLINK=-dynamic -bundle -undefined suppress -force_flat_namespace
else
SOLINK=-shared -Xlinker -x
endif
#
# targets
#
all: $(SHAREDOS)
clean:
rm -f *.so *.o $(OBJS)
app_conference.so : $(OBJS)
$(CC) -pg -shared -Xlinker -x -o $@ $(OBJS)
vad_test: vad_test.o libspeex/preprocess.o libspeex/misc.o
libspeex/smallft.o
$(CC) $(PROFILE) -o $@ $^ -lm
install: all
for x in $(SHAREDOS); do $(INSTALL) -m 755 $$x
$(INSTALL_MODULES_DIR) ; done
cp -f *.gsm /var/lib/asterisk/sounds
cp -f *.wav /var/lib/asterisk/sounds
# /var/horizon/mojo/lib/horizoncmd restart asterisk
# make sure you restart asterisk after make install
# config: all
# cp conf.conf /etc/asterisk/
*end of makefile
On 6/4/06, Ricardo Martins [EMAIL PROTECTED] wrote:
Do anybody could compile app_conference on x86_64??? I tryied with two
versions of app_conference and got the same problem on compiling:
relocation R_X86_64_32 against `a local symbol' can not be used when
making a shared recompile with -fPIC
app_conference.o: could not read symbols: Bad value
ENVIRONMENT:
---
Machine: DELL PE-2850 with two processors Xeon 3.0GHz
Kernel: 2.6.9-34.0.1.ELsmp
Version of app_conference (Both):
http://www.eflo.net/files/VD_app_conference_0.4.zip; or
http://www.eflo.net/files/app_conference.tar.gz;
---
COMPLETE output of compilation:
--
gcc -pipe -Wall -Wmissing-prototypes -Wmissing-declarations -g
-I/root/local/asterisk/asterisk/include
[Asterisk-Users] syslog server
Does anyone know a good syslog server to use for grandstream phones? I want to set this up to see what is happening with the grandstreams. Easy and Free preferably. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] syslog server
hmmm... I am a huge fan of syslog-ng, but the stock syslog on your * system should work well... Matthew Warren wrote: Does anyone know a good syslog server to use for grandstream phones? I want to set this up to see what is happening with the grandstreams. Easy and Free preferably. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Change in dial command behaviour between 1.2.7.1 and 1.2.8?
On Tue, 6 Jun 2006, Michiel van Baak wrote: On 15:46, Tue 06 Jun 06, Remco Barendse wrote: On Tue, 6 Jun 2006, Kevin P. Fleming wrote: - Remco Barendse [EMAIL PROTECTED] wrote: Did I goof up or did something change? No, there should not be any behavioral changes between 1.2.7.1 and 1.2.8 except for bug fixes. The only change that I see in the ChangeLog for 1.2.8 that could be relevant is the one regarding 'usecallingpres' handling chan_zap; are you using that option in your zapata.conf file? Yes I do have usecallingpres=yes in my zaptel.conf. I already noticed this and tried commenting out this line and did a reload on * but that didn't change much. I'll try downgrading zaptel first, if it doesn't work I will also downgrade asterisk back to 1.2.7.1 Try a restart first. A reload may not be enough. Restarting asterisk is easier than reloading zaptel, I tried downgrading to Asterisk 1.2.7.1 but there is no change. I will try downgrading to zaptel tonight. Thanks! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Compiling QuaBri cards
Hello, I've installed a QuadBri card from Junghanns, and have some problems for compiling software. Compiling Zaptel and LibPRI are OK. But when i want to compile ztgsm files, i have the error: link /usr/src/linux-2.6 to your kernel sources first ! I work with kernel 2.6.8-2-686, and i have a link: linux-2.6.8-2-686 from /lib/modules/2.6.8-2-686. When i creat a link for linux-2.6 on this directory, i have some news erreors; has: structure has no member... Could you help me on this problem?? Best regards, -- Olivier Saulnier STEGANUX 1er étage Diamecans Bel Air 03410 St Victor T: 04.70.02.27.62 F: 04.70.09.97.41 http://www.steganux.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Playback welcome message while phones ring, please help
Yes it does, I just set our system up that way. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Gareth Blades Sent: Tuesday, June 06, 2006 6:37 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Playback welcome message while phones ring,please help I believe if you use the new native music on hold feature it always plays the music on hold starting from the beginning. On Tue, 2006-06-06 at 11:15, Tommaso Calosi wrote: I want that incoming callers to hear a welcome message while the phones ring. I know I can use Dial with the m(class) option to make the same with musiconhold, but the problem is that musiconhold does not start from the beginning of my mp3 file. If I use Playback or Background, the phones do not ring unless the mp3 file is over... Any suggestion? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] syslog server
On 11:02, Tue 06 Jun 06, Matthew Warren wrote: Does anyone know a good syslog server to use for grandstream phones? I want to set this up to see what is happening with the grandstreams. Easy and Free preferably. Hi, Use the syslog on your asterisk box :) -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called users? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk exit on startup
I'm having a problem with a new installation of asterisk 1.2.5 with a digium dual port T1 (span 1 connected to an outside line, and span 2 connected to a CAC access bank I channel bank with 24 fxs ports). When I start Asterisk (either from safe_asterisk or asterisk -vvvc) it will immediately exit after it initializes. It will start the logger, register applications and functions, register all 48 channels, load all the .so modules, and then exit. The logs complain about not having dundi.conf and iax.conf, but as I will not be using dundi or iax at the moment, I don't see why it should need those particular conf files and not others (like sip.conf and manager.conf). If anyone has experienced a similar problem, or can point me in the correct direction to get rolling, it'd be greatly appreciated. Here's all the log files I'm currently using (all in /etc/asterisk except zaptel.conf): --- zaptel.conf --- span=1,1,0,esf,b8zs span=2,0,0,esf,b8zs fxoks=1-24 fxsls=25-48 loadzone = us defaultzone=us --- zapata.conf --- [channels] usecallerid=yes hidecallerid=no transfer=yes echocancel=yes echotraining=yes immediate=no ; span 1 (outside T1) context=incoming signalling=fxo_ks channel = 1-24 ; span 2 (internal fxs ports) context=internal signalling=fxs_ls channel = 25-48 --- logger.conf --- [general] [logfiles] verbose_log = verbose messages = notice,warning,error debug = debug all_log = debug,notice,verbose,warning --- extensions.conf --- ; test context for internal analog phones [internal] exten = _X.,1,Playback(demo-thanks) ; test context for incoming T1 calls [incoming] exten = _X.,1,Playback(demo-thanks) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] syslog server
www.kiwisyslog.com works perfect but windows only hth -Original Message- From: Matthew Warren [mailto:[EMAIL PROTECTED] Sent: Tuesday, June 06, 2006 9:03 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] syslog server Does anyone know a good syslog server to use for grandstream phones? I want to set this up to see what is happening with the grandstreams. Easy and Free preferably. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Playback welcome message while phones ring, please help
2006/6/6, Gareth Blades [EMAIL PROTECTED]: I believe if you use the new native music on hold feature it alwaysplays the music on hold starting from the beginning.Where can I find this new native music on hold feature ? In Asterisk 1.2.x ?Regards ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Vonage and FXO
Is anyone using Vonage on an FXO port in Asterisk? How well does it work? Specifically, any echo/delay problems? Second part, I am assuming it is possible to separate fxo ports for least cost routing correct? In other words, I would like the routing to be such that any local or 1-800# dials fxo ports 1-4 which is a normal 4 line analog PSTN connection. Any long distance call will try to dial fxo port 5 (Vonage ATA) first and if it's used then use fxo ports 1-4. Is this easy to do in FreePBX? I know I can get a Vonage softphone account and not use an ATA/FXO port. I want to know if I can do it with an ATA/FXO. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Compiling QuaBri cards
On Tue, Jun 06, 2006 at 05:27:58PM +0200, Olivier Saulnier wrote: Hello, I've installed a QuadBri card from Junghanns, and have some problems for compiling software. Compiling Zaptel and LibPRI are OK. But when i want to compile ztgsm files, i have the error: link /usr/src/linux-2.6 to your kernel sources first ! I work with kernel 2.6.8-2-686, and i have a link: linux-2.6.8-2-686 from /lib/modules/2.6.8-2-686. When i creat a link for linux-2.6 on this directory, i have some news erreors; has: structure has no member... Could you help me on this problem?? Best regards, wget http://rapid.dotsrc.org/rapid/pool/main/z/zaptel/zaptel-modules-2.6.8-2-686_1.2.5-4+2.6.8-16sarge1_i386.deb Alternatively, install zaptel-source from there and build using m-a a-i zaptel -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 [EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Compile install error.
Ok I tried the 1.2.7 install again and I get the same result. I do not remember that from when I installed it the first time. Obviously my shell does not like something in the context that checks for old modules. At first I thought it was because there were old modules there but deleting them makes no difference. Doug On Tue, 6 Jun 2006, Kevin P. Fleming wrote: - Doug Crompton [EMAIL PROTECTED] wrote: I am getting the following error at the end of 'make install' 1.2.9 I have not tried to find it but I suspect there is just a misplaced punctuation. It runs fine. That part of the Makefile has not been touched in quite a while, so there should not be any behavioral differences from 1.2.8, 1.2.7.1, etc. If this is repeatable on your system, please open a bug report on bugs.digium.com. -- Kevin P. Fleming Senior Software Engineer Digium, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Those that sacrifice essential liberty to obtain a little temporary safety deserve neither liberty nor safety. -- Ben Franklin (1759) * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307* * * * [EMAIL PROTECTED]* * http://www.crompton.com * ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk exit on startup
On Tue, Jun 06, 2006 at 10:10:17AM -0600, Nathan Bell wrote: I'm having a problem with a new installation of asterisk 1.2.5 with a digium dual port T1 (span 1 connected to an outside line, and span 2 connected to a CAC access bank I channel bank with 24 fxs ports). When I start Asterisk (either from safe_asterisk or asterisk -vvvc) it will immediately exit after it initializes. It will start the logger, register applications and functions, register all 48 channels, load all the .so modules, and then exit. What exactly is the error message about? Could you attach the logs? The logs complain about not having dundi.conf and iax.conf, but as I will not be using dundi or iax at the moment, I don't see why it should need those particular conf files and not others (like sip.conf and manager.conf). I figure that those are warnings. If anyone has experienced a similar problem, or can point me in the correct direction to get rolling, it'd be greatly appreciated. Here's all the log files I'm currently using (all in /etc/asterisk except zaptel.conf): --- zaptel.conf --- span=1,1,0,esf,b8zs span=2,0,0,esf,b8zs fxoks=1-24 fxsls=25-48 loadzone = us defaultzone=us --- zapata.conf --- [channels] usecallerid=yes hidecallerid=no transfer=yes echocancel=yes echotraining=yes immediate=no ; span 1 (outside T1) context=incoming signalling=fxo_ks channel = 1-24 ; span 2 (internal fxs ports) context=internal signalling=fxs_ls channel = 25-48 --- logger.conf --- [general] [logfiles] verbose_log = verbose messages = notice,warning,error debug = debug all_log = debug,notice,verbose,warning --- extensions.conf --- ; test context for internal analog phones [internal] exten = _X.,1,Playback(demo-thanks) ; test context for incoming T1 calls [incoming] exten = _X.,1,Playback(demo-thanks) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 [EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ISDN BRI (I.430) over ethernet
2006/6/6, Woodoo People .pGa! [EMAIL PROTECTED]: CPE-Asterisk(NT Mode)-ip-Asterisk(CPE)-NT?Have you already tried such setup ?What are the benefits of using Asterisk instead a dedicated CPE ?regards ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Vonage and FXO
I used a scenario like this before but I always ran into intermittent echo issues that were just not worth the hassle for me so I switched to a sole IP origination and termination service. Just my personal experience! HTH Curt -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of mustardman29 Sent: Tuesday, June 06, 2006 12:22 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Vonage and FXO Is anyone using Vonage on an FXO port in Asterisk? How well does it work? Specifically, any echo/delay problems? Second part, I am assuming it is possible to separate fxo ports for least cost routing correct? In other words, I would like the routing to be such that any local or 1-800# dials fxo ports 1-4 which is a normal 4 line analog PSTN connection. Any long distance call will try to dial fxo port 5 (Vonage ATA) first and if it's used then use fxo ports 1-4. Is this easy to do in FreePBX? I know I can get a Vonage softphone account and not use an ATA/FXO port. I want to know if I can do it with an ATA/FXO. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DTMF feedthru again...
Ok trying this again... is there anyone using the SPA-3000 with * I am not sure if this is a specific problem to it or not. This is something I really need to fix!!! When dialing out using * interfaced to an SPA-3000, fxs,fxo, I cannot access (reliably) DTMF menus at the called party, after call completion. Dialing DTMF is fine. I checked by calling myself. Listening to either end on a completed call, and pressing a DTMF button on the opposing phone results in an audible click and very little if any audible DTMF energy being heard. What is muting the DTMF??? Does * have anything to do with this? I am not using any dial flags. I tried 'inband' in all places with no difference. At one point this seemed like a * feature problem and I thought removing dial flags fixed it but that does not now seem to be the case. How can I fix this??? Doug * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307* * * * [EMAIL PROTECTED]* * http://www.crompton.com * ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Change in dial command behaviour between 1.2.7.1 and 1.2.8?
On Tue, 6 Jun 2006, Kevin P. Fleming wrote: - Remco Barendse [EMAIL PROTECTED] wrote: Did I goof up or did something change? No, there should not be any behavioral changes between 1.2.7.1 and 1.2.8 except for bug fixes. The only change that I see in the ChangeLog for 1.2.8 that could be relevant is the one regarding 'usecallingpres' handling chan_zap; are you using that option in your zapata.conf file? Found it!! In zapata.conf I removed 2 lines ages ago : pridialplan=unknown prilocaldialplan=unknown For some reason when restarting / reloading Asterisk this never made any difference, only now this problem popped up. So in the end it turned out that I did goof up :) Why this setting is preventing proper DID notification to be sent to the PBX is a mystery to me but it seems to work now. Thanks for all the replies! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] wav49 size for a 3 minute voicemail
Hi, I tried to find a reference in terms of size but got back a bunch of tech documents and couldn't get the idea of wav49 format. wav49 format is supposed to be half the size of a normal wav right? so, how much disk space takes to save one minute of audio in wav49? I trying to do some capacity planning for a voicemail server. -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 mailto: [EMAIL PROTECTED] mailto: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk 1.2.9.1 and 1.0.11.1 Released -- Security Fix
The Asterisk Development Team today re-released Asterisk 1.2.9.1 and Asterisk 1.0.11.1 to address a security vulnerability in the IAX2 channel driver (chan_iax2). The vulnerability affects all users with IAX2 clients that might be compromised or used by a malicious user, and can lead to denial of service attacks and random Asterisk server crashes via a relatively trivial exploit. These re-releases correct a problem introduced by the vulnerability fix involving transport of video frames over IAX2. All users are urged to upgrade as soon as they can practically do so, or ensure that they don't expose IAX2 services to the public if it is not necessary. The release files are available in the usual place (ftp.digium.com), as both tarballs and patch files relative to the last release. In addition, both the tarballs and the patch files have been signed using GPG keys of the release maintainers, so that you can ensure their authenticity. Thank you for your support of Asterisk! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP One-way audio: == Forcing Marker bit, because SSRC has changed - trxtel.com
Dear list (and more specifically Bret), I am getting one-way (inbound only) audio when trying to place a SIP call via voip.trxtel.com (i.e. [EMAIL PROTECTED]). The Cli spits out == Forcing Marker bit, because SSRC has changed 5 times after atempting a native bridge. I realize this is most certainly a NAT issue, the * server is behind one. Sip.conf has externip=, and localnet=. Can someone explain what SSRC changing implies is going on, and why it affects NAT? This is actually the first NAT issue I have had with *, all other SIP calls have worked fine (I must be lucky). Sincerely, Brent A. Torrenga Torrenga Engineering, Inc. 907 Ridge Road Munster, Indiana 46321-1771 tel:+1 219 836 8918 x325 fax:+1 219 836 1138 email:[EMAIL PROTECTED] web:www.torrenga.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Playback welcome message while phones ring, please help
I am running on 1.2.7.1 -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of OlivierSent: Tuesday, June 06, 2006 12:17 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Playback welcome message while phones ring,please help 2006/6/6, Gareth Blades [EMAIL PROTECTED]: I believe if you use the new native music on hold feature it alwaysplays the music on hold starting from the beginning.Where can I find this "new native music on hold feature" ?In Asterisk 1.2.x ?Regards ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk exit on startup
The errors were just these: Jun 6 10:03:06 ERROR[14553] pbx_dundi.c: Unable to load config dundi.conf Jun 6 10:03:06 ERROR[14553] chan_iax2.c: Unable to load config iax.conf There were other warnings and notices for the other conf files, and that was it. However, I just noticed a brand spankin' new version of Asterisk (1.2.9), and it fixes the problem. Sorry for the inconvenience, but thanks for the help. Tzafrir Cohen wrote: On Tue, Jun 06, 2006 at 10:10:17AM -0600, Nathan Bell wrote: I'm having a problem with a new installation of asterisk 1.2.5 with a digium dual port T1 (span 1 connected to an outside line, and span 2 connected to a CAC access bank I channel bank with 24 fxs ports). When I start Asterisk (either from safe_asterisk or asterisk -vvvc) it will immediately exit after it initializes. It will start the logger, register applications and functions, register all 48 channels, load all the .so modules, and then exit. What exactly is the error message about? Could you attach the logs? The logs complain about not having dundi.conf and iax.conf, but as I will not be using dundi or iax at the moment, I don't see why it should need those particular conf files and not others (like sip.conf and manager.conf). I figure that those are warnings. If anyone has experienced a similar problem, or can point me in the correct direction to get rolling, it'd be greatly appreciated. Here's all the log files I'm currently using (all in /etc/asterisk except zaptel.conf): --- zaptel.conf --- span=1,1,0,esf,b8zs span=2,0,0,esf,b8zs fxoks=1-24 fxsls=25-48 loadzone = us defaultzone=us --- zapata.conf --- [channels] usecallerid=yes hidecallerid=no transfer=yes echocancel=yes echotraining=yes immediate=no ; span 1 (outside T1) context=incoming signalling=fxo_ks channel = 1-24 ; span 2 (internal fxs ports) context=internal signalling=fxs_ls channel = 25-48 --- logger.conf --- [general] [logfiles] verbose_log = verbose messages = notice,warning,error debug = debug all_log = debug,notice,verbose,warning --- extensions.conf --- ; test context for internal analog phones [internal] exten = _X.,1,Playback(demo-thanks) ; test context for incoming T1 calls [incoming] exten = _X.,1,Playback(demo-thanks) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Vonage and FXO
I am using it with my [EMAIL PROTECTED] setup. I did not face any issues with echo, but once in a while, the trunk does NOT get disconnected even after the call has been completed. So I had to manually plug the phone cable out from FXO and plug it back again. But I think that's something to do with my version of FXO drivers. Otherwise it works for me. Paddu -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Curt Shaffer Sent: Tuesday, June 06, 2006 10:37 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Vonage and FXO I used a scenario like this before but I always ran into intermittent echo issues that were just not worth the hassle for me so I switched to a sole IP origination and termination service. Just my personal experience! HTH Curt -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of mustardman29 Sent: Tuesday, June 06, 2006 12:22 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Vonage and FXO Is anyone using Vonage on an FXO port in Asterisk? How well does it work? Specifically, any echo/delay problems? Second part, I am assuming it is possible to separate fxo ports for least cost routing correct? In other words, I would like the routing to be such that any local or 1-800# dials fxo ports 1-4 which is a normal 4 line analog PSTN connection. Any long distance call will try to dial fxo port 5 (Vonage ATA) first and if it's used then use fxo ports 1-4. Is this easy to do in FreePBX? I know I can get a Vonage softphone account and not use an ATA/FXO port. I want to know if I can do it with an ATA/FXO. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DTMF feedthru again...
try setting dtmf playback length to .5 in the admin section of the Sipura and try again. On 6/6/06, Doug Crompton [EMAIL PROTECTED] wrote: Ok trying this again... is there anyone using the SPA-3000 with * I am not sure if this is a specific problem to it or not. This is something I really need to fix!!! When dialing out using * interfaced to an SPA-3000, fxs,fxo, I cannot access (reliably) DTMF menus at the called party, after call completion. Dialing DTMF is fine. I checked by calling myself. Listening to either end on a completed call, and pressing a DTMF button on the opposing phone results in an audible click and very little if any audible DTMF energy being heard. What is muting the DTMF??? Does * have anything to do with this? I am not using any dial flags. I tried 'inband' in all places with no difference. At one point this seemed like a * feature problem and I thought removing dial flags fixed it but that does not now seem to be the case. How can I fix this??? Doug * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307* * * * [EMAIL PROTECTED]* * http://www.crompton.com * ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- 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 provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Playback welcome message while phones ring, please help
Thanks, it works for me too. Tim Sharp wrote: Yes it does, I just set our system up that way. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Gareth Blades Sent: Tuesday, June 06, 2006 6:37 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Playback welcome message while phones ring,please help I believe if you use the new native music on hold feature it always plays the music on hold starting from the beginning. On Tue, 2006-06-06 at 11:15, Tommaso Calosi wrote: I want that incoming callers to hear a welcome message while the phones ring. I know I can use Dial with the m(class) option to make the same with musiconhold, but the problem is that musiconhold does not start from the beginning of my mp3 file. If I use Playback or Background, the phones do not ring unless the mp3 file is over... Any suggestion? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom SIP 1.6.6
I'd suggest calling whoever you buy your phones from. The distributor I work with requires that you are Polycom certified to be able to purchase phones from them, but once you are certified with Polycom you can actually download the firmware from their extranet. On 6/5/06, Douglas Garstang [EMAIL PROTECTED] wrote: Off topic. Anyone know where I can get Polycom SIP software v1.6.6, unofficially?Not that Polycom is analy retentive, or anything like that... Doug___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP One-way audio: == Forcing Marker bit, because SSRC has changed - trxtel.com
- Brent Torrenga [EMAIL PROTECTED] wrote: == Forcing Marker bit, because SSRC has changed 5 times after atempting a native bridge. I realize this is most certainly a NAT issue, the * server is behind one. Sip.conf has externip=, and localnet=. google is your friend :-) We've already covered this on the list, it's not something that you should be concerned about (or even seeing). Later releases of Asterisk no longer generate this message, and when they do it is not marked as a warning, because it is not. -- Kevin P. Fleming Senior Software Engineer Digium, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Weird Can-Reinvite problem
Hi All, I'm having a really weird can reinvite issue. I've been banging my head around on this for days now.. I have two asterisk servers. One at 172.20.0.11 One at 172.20.2.5 172.20.0.11 is a hosted box and serves multiple offices 172.20.2.5 is a box on site at a customer's office. A phone at 172.20.128.10 makes a call using server 172.20.0.11 to a phone at 172.20.2.80 via server 172.20.2.5: Phone A--asterisk A-SER-asterisk B---PhoneB All devices all have ip connectivity (No Firewalls! No Natting) to each other. so phone a can ping phone b and server b, etc, etc, etc.. Can reinvite is enabled on both the ser connection (on both sides) and for both phones.. Making a call from phone A to phone B works great.. Except you can hear a pop when the reinvite happens. After the call is connected Phone B can transfer the phone just fine.. However if phone A (the originator) tries to transfer FIRST (either to the pstn via SER or to another local extension on asterisk A) the call will have 0 way audio. If the call is transfered back, there will be one way audio. It seems this is Always how it is, over and over.. The Originator Cannot transfer the call first. I THINK if the destination transfers first, THEN the originator can transfer.. I've checked netmasks, ips, gateways, etc, etc.. The SDP on the reinvites looks ok.. No Nat, no funny business here.. just IP routing.. Any ideas? -Brett ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Vonage and FXO
It would be helpful if responders would tell us what FXO hardware they are using and which vonage ATA device it connects to. Padmanaban Balasubramaniam wrote: I am using it with my [EMAIL PROTECTED] setup. I did not face any issues with echo, but once in a while, the trunk does NOT get disconnected even after the call has been completed. So I had to manually plug the phone cable out from FXO and plug it back again. But I think that's something to do with my version of FXO drivers. Otherwise it works for me. Paddu -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Curt Shaffer Sent: Tuesday, June 06, 2006 10:37 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Vonage and FXO I used a scenario like this before but I always ran into intermittent echo issues that were just not worth the hassle for me so I switched to a sole IP origination and termination service. Just my personal experience! HTH Curt -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of mustardman29 Sent: Tuesday, June 06, 2006 12:22 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Vonage and FXO Is anyone using Vonage on an FXO port in Asterisk? How well does it work? Specifically, any echo/delay problems? Second part, I am assuming it is possible to separate fxo ports for least cost routing correct? In other words, I would like the routing to be such that any local or 1-800# dials fxo ports 1-4 which is a normal 4 line analog PSTN connection. Any long distance call will try to dial fxo port 5 (Vonage ATA) first and if it's used then use fxo ports 1-4. Is this easy to do in FreePBX? I know I can get a Vonage softphone account and not use an ATA/FXO port. I want to know if I can do it with an ATA/FXO. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DTMF feedthru again...
Tried that makes no difference. Did it for you? What DMF method(s) are you using. Looking at a goggle search yields lots of talk on this but no real solution. Apparently there is an rfc2833 issue and * is working on it??? Also it appears the codec used might be an issue. This is a serious problem in my book. It precludes me from using any DTMF over PSTN with * at this point. Any further help or explanation would be appreciated. On Tue, 6 Jun 2006, Tom Vile wrote: try setting dtmf playback length to .5 in the admin section of the Sipura and try again. On 6/6/06, Doug Crompton [EMAIL PROTECTED] wrote: Ok trying this again... is there anyone using the SPA-3000 with * I am not sure if this is a specific problem to it or not. This is something I really need to fix!!! When dialing out using * interfaced to an SPA-3000, fxs,fxo, I cannot access (reliably) DTMF menus at the called party, after call completion. Dialing DTMF is fine. I checked by calling myself. Listening to either end on a completed call, and pressing a DTMF button on the opposing phone results in an audible click and very little if any audible DTMF energy being heard. What is muting the DTMF??? Does * have anything to do with this? I am not using any dial flags. I tried 'inband' in all places with no difference. At one point this seemed like a * feature problem and I thought removing dial flags fixed it but that does not now seem to be the case. How can I fix this??? Doug * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307* * * * [EMAIL PROTECTED]* * http://www.crompton.com * ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- 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 provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Those that sacrifice essential liberty to obtain a little temporary safety deserve neither liberty nor safety. -- Ben Franklin (1759) * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307* * * * [EMAIL PROTECTED]* * http://www.crompton.com * ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DTMF feedthru again...
Also to expand on this... when listening to opposing phone in a connected call over PSTN you hear a click followed by a very short burst of DTMF audible energy. Same in both directions. I can't be the only one having this problem! Doug On Tue, 6 Jun 2006, Tom Vile wrote: try setting dtmf playback length to .5 in the admin section of the Sipura and try again. On 6/6/06, Doug Crompton [EMAIL PROTECTED] wrote: Ok trying this again... is there anyone using the SPA-3000 with * I am not sure if this is a specific problem to it or not. This is something I really need to fix!!! When dialing out using * interfaced to an SPA-3000, fxs,fxo, I cannot access (reliably) DTMF menus at the called party, after call completion. Dialing DTMF is fine. I checked by calling myself. Listening to either end on a completed call, and pressing a DTMF button on the opposing phone results in an audible click and very little if any audible DTMF energy being heard. What is muting the DTMF??? Does * have anything to do with this? I am not using any dial flags. I tried 'inband' in all places with no difference. At one point this seemed like a * feature problem and I thought removing dial flags fixed it but that does not now seem to be the case. How can I fix this??? Doug * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307* * * * [EMAIL PROTECTED]* * http://www.crompton.com * ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- 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 provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Those that sacrifice essential liberty to obtain a little temporary safety deserve neither liberty nor safety. -- Ben Franklin (1759) * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307* * * * [EMAIL PROTECTED]* * http://www.crompton.com * ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Transcoding g.711 - g.729
Hello, I have an asterisk server running with 23 g.729 licenses. I have also purchased a sound file from thevoice.digium.com. I need to covert this file (uLaw, PCM I think) to g.711, g.729 g.723 for use with an IVR system. Is there a way I can convert the files using the g.729 digium codec? sox? Thanks -Matt -- Matthew S. Crocker Vice President Crocker Communications, Inc. Internet Division PO BOX 710 Greenfield, MA 01302-0710 http://www.crocker.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DTMF feedthru again...
Using AVT in my sipura with above settings and it work fine going out the PSTN. There was an issue a while back with an older version of Asterisk with one of my providers but it has been fine since the upgrade. I also use ulaw for calls. On 6/6/06, Doug Crompton [EMAIL PROTECTED] wrote: Tried that makes no difference. Did it for you? What DMF method(s) are you using. Looking at a goggle search yields lots of talk on this but no real solution. Apparently there is an rfc2833 issue and * is working on it??? Also it appears the codec used might be an issue. This is a serious problem in my book. It precludes me from using any DTMF over PSTN with * at this point. Any further help or explanation would be appreciated. On Tue, 6 Jun 2006, Tom Vile wrote: try setting dtmf playback length to .5 in the admin section of the Sipura and try again. On 6/6/06, Doug Crompton [EMAIL PROTECTED] wrote: Ok trying this again... is there anyone using the SPA-3000 with * I am not sure if this is a specific problem to it or not. This is something I really need to fix!!! When dialing out using * interfaced to an SPA-3000, fxs,fxo, I cannot access (reliably) DTMF menus at the called party, after call completion. Dialing DTMF is fine. I checked by calling myself. Listening to either end on a completed call, and pressing a DTMF button on the opposing phone results in an audible click and very little if any audible DTMF energy being heard. What is muting the DTMF??? Does * have anything to do with this? I am not using any dial flags. I tried 'inband' in all places with no difference. At one point this seemed like a * feature problem and I thought removing dial flags fixed it but that does not now seem to be the case. How can I fix this??? Doug * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307* * * * [EMAIL PROTECTED]* * http://www.crompton.com * ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- 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 provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Those that sacrifice essential liberty to obtain a little temporary safety deserve neither liberty nor safety. -- Ben Franklin (1759) * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307* * * * [EMAIL PROTECTED]* * http://www.crompton.com * ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- 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 provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX Passing Variables
Well, this kinda sux. We have three Asterisk servers. Phones register to a single, primary server. When a phone on one wants to reach a phone on another, we use DUNDi to discover the destination pbx and IAX to transfer the call to the other Asterisk box. This seems to be a fairly common practice amongst Asterisk users, yes? Well, what about setting variables before call placement? Say you want to set the variable _ALERT_INFO, to have Polycom phones auto answer? Essentially the problem is that channel variables (with the exception of caller id) are not passed from one Asterisk box to another with IAX. How have people gotten around this? Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: fine-tuning asterisk questions
William Piper wrote:
For Problem #2:
I'm not sure what you are asking. Perhaps post your dialplan for this
problem we will take a look.
bp
On 6/4/06, *M.Hockings* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
Problem 2) Incoming sip calls from my voip provider get rejected unless
I allow anyone to connect with sip. I have an incoming route set up with
the right DID that matches the DID that asterisk picks out but it still
rejects the call. Any suggestions about how to get this to work without
allowing any sip connection?
Mike
Hi William, at the bottom of this is my extensions.conf which seems to
be the largest part of the equation for problem #2. I have not applied
any changes to try and resolve my problem #1 yet.
I think the question here is the operation of the following statement in
the [from-sip-external] section:
exten = s,1,GotoIf($[${ALLOW_SIP_ANON}=yes]?from-trunk,${DID},1)
If I interpret it correctly it should go to from-trunk,1 if the freePBX
allow anonymous sip connections is true and go to
incoming-sip-did-value,1 if it is false ? That is should I be looking
for something like this in the config files to understand how this would
be handled?
exten=416967,1,
As an aside, is there some beginners guide to understanding dial plans?
My original dial plan (based on things read on voip-info.org) was very
simple and worked as far as it was configured. I have recently gone to
freePBX to try and make the dial plan changes easier and faster however
it adds a lot of gorp like this that I don't understand.
Thanks for any guidance on this,
Mike
; Asterisk Management Portal (AMP)
; Copyright (C) 2004 Coalescent Systems Inc
; dialparties.agi (http://www.sprackett.com/asterisk/)
; Asterisk::AGI (http://asterisk.gnuinter.net/)
; gsm (http://www.ibiblio.org/pub/Linux/utils/compress/!INDEX.short.html)
; loligo sounds (http://www.loligo.com/asterisk/sounds/)
; mpg123 (http://voip-info.org/wiki-Asterisk+config+musiconhold.conf)
; include extension contexts generated from AMP
#include extensions_additional.conf
; Customizations to this dialplan should be made in extensions_custom.conf
; See extensions_custom.conf.sample for an example
#include extensions_custom.conf
[from-trunk]; just an alias
since VoIP shouldn't be called PSTN
include = from-pstn
[from-pstn]
include = from-pstn-custom ; create this context in
extensions_custom.conf to include customizations
include = ext-findmefollow ; MODIFICATOIN (PL) for findmefollow if
enabled, should be bofore ext-local
include = ext-did-direct ; MODIFICATOIN (PL) put before ext-did to
take precedence
include = ext-did
exten = fax,1,Goto(ext-fax,in_fax,1)
; MODIFICATION (PL)
;
; Required to assure that direct dids go to personal ring group before
local extension.
; This could be auto-generated however I it is prefered to be put here
and hard coded
; so that it can be modified if ext-local should take precedence in
certain situations.
; will have to decide what to do later.
;
[from-did-direct]
include = ext-findmefollow
include = ext-local
;
; Macros [macro]
;
; Rings one or more extensions. Handles things like call forwarding and DND
; We don't call dial directly for anything internal anymore.
; ARGS: $TIMER, $OPTIONS, $EXT1, $EXT2, $EXT3, ...
; Use a Macro call such as the following:
; Macro(dial,$DIAL_TIMER,$DIAL_OPTIONS,$EXT1,$EXT2,$EXT3,...)
[macro-dial]
exten = s,1,AGI,dialparties.agi
exten = s,2,NoOp(Returned from dialparties with no extensions to call)
exten = s,3,NoOp(DIALSTATUS is '${DIALSTATUS}')
exten = s,10,Dial(${ds}) ; dialparties
will set the priority to 10 if $ds is not null
exten = s,20,NoOp(Returned from dialparties with hunt groups to dial )
exten = s,21,Set(HuntLoop=0)
exten = s,22,GotoIf($[${HuntMembers} = 1]?30 ) ; if this is from
rg-group, don't strip prefix
exten = s,23,NoOp(Returning there are no members left in the hunt group
to ring)
exten = s,30,Set(HuntMember=HuntMember${HuntLoop})
exten = s,31,GotoIf($[$[${CALLTRACE_HUNT} != ]
$[${RingGroupMethod} = hunt ]]?32:35 ) ; Set CAll Trace for Hunt
member we are going to call
exten = s,32,Set(CT_EXTEN=${CUT(ARG3,,$[${HuntLoop} + 1])})
exten = s,33,Set(DB(CALLTRACE/${CT_EXTEN})=${CALLTRACE_HUNT})
exten = s,34,Goto(s,42)
exten = s,35,GotoIf($[$[${CALLTRACE_HUNT} != ]
$[${RingGroupMethod} = memoryhunt ]]?36:50 ) ;Set Call Trace for
each hunt member we are going to call Memory groups have multiple
members to set CALL TRACE For hence the loop
exten = s,36,Set(CTLoop=0)
exten = s,37,GotoIf($[${CTLoop} ${HuntLoop}]?42 ) ; if this is from
rg-group, don't strip prefix
exten = s,38,Set(CT_EXTEN=${CUT(ARG3,,$[${CTLoop} + 1])})
Re: [Asterisk-Users] Weird Can-Reinvite problem
Have you tried turning off icmp redirect on your router?On 6/6/06, Brett N [EMAIL PROTECTED] wrote: Hi All,I'm having a really weird can reinvite issue. I've been banging my head around on this for days now..I have two asterisk servers. One at 172.20.0.11 One at 172.20.2.5172.20.0.11 is a hosted box and serves multiple offices172.20.2.5 is a box on site at a customer's office.A phone at 172.20.128.10 makes a call using server 172.20.0.11 to a phoneat 172.20.2.80 via server 172.20.2.5:Phone A--asterisk A-SER-asterisk B---PhoneB All devices all have ip connectivity (No Firewalls! No Natting) to eachother. so phone a can ping phone b and server b, etc, etc, etc..Can reinvite is enabled on both the ser connection (on both sides) and for both phones..Making a call from phone A to phone B works great.. Except you can hear apop when the reinvite happens. After the call is connected Phone B cantransfer the phone just fine.. However if phone A (the originator) tries to transfer FIRST (either to the pstn via SER or to another localextension on asterisk A) the call will have 0 way audio. If the call istransfered back, there will be one way audio.It seems this is Always how it is, over and over.. The Originator Cannot transfer the call first. I THINK if the destination transfers first, THENthe originator can transfer..I've checked netmasks, ips, gateways, etc, etc.. The SDP on the reinviteslooks ok..No Nat, no funny business here.. just IP routing.. Any ideas?-Brett___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Jim Freeze ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] OT: Cellular boosters
We use Motorola v551's as extensions on our Asterisk system with a homebrew find me/follow me dialplan. It works great except where coverage is poor then of course the inbound call hits voicemail. This has nothing to do with Asterisk and everything to do with our cellular provider, but since you guys are telephony pros I'd like to ask if anyone has had any good or bad experience with gain boosters for cells from those snake oil stick on things all the way up to powered one-watt boosters. Ideally, I'd like a situation where I replace the stock OEM antenna with something else for $10 and away I go. I have a hundred guys with v551s that are pissed about missed calls, so any and all suggestions are welcome. tia ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DTMF feedthru again...
in Fact we saw similar problems with all sipura products. We think its a default value thats not quite right for the north american market, these units are built and tested in asia mostly.one simple test to check it out is call this number
www.nextwavetitaniumplus.com Toll-Free Account Information Line: 888-252-9535it just seemd that even the cisco is not passing the dtmf ..Can anyone confirm ?
On 6/6/06, Doug Crompton [EMAIL PROTECTED] wrote:
Also to expand on this... when listening to opposing phone in a connectedcall over PSTN you hear a click followed by a very short burst of DTMFaudible energy. Same in both directions.I can't be the only one having this problem!
DougOn Tue, 6 Jun 2006, Tom Vile wrote: try setting dtmf playback length to .5 in the admin section of the Sipura and try again. On 6/6/06, Doug Crompton
[EMAIL PROTECTED] wrote: Ok trying this again... is there anyone using the SPA-3000 with * I am not sure if this is a specific problem to it or not. This is something I really need to fix!!!
When dialing out using * interfaced to an SPA-3000, fxs,fxo, I cannot access (reliably) DTMF menus at the called party, after call completion. Dialing DTMF is fine.
I checked by calling myself. Listening to either end on a completed call, and pressing a DTMF button on the opposing phone results in an audible click and very little if any audible DTMF energy being heard.
What is muting the DTMF??? Does * have anything to do with this? I am not using any dial flags. I tried 'inband' in all places with no difference. At one point this
seemed like a * feature problem and I thought removing dial flags fixed it but that does not now seem to be the case. How can I fix this??? Doug
*Doug Crompton * *Richboro, PA 18954* *215-431-6307* **
* [EMAIL PROTECTED]* * http://www.crompton.com*
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-- 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 provided by Easynews.com -- Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-usersThose that sacrifice essential liberty to obtain a little temporary safety
deserve neither liberty nor safety.-- Ben Franklin (1759)*Doug Crompton **Richboro, PA 18954**215-431-6307***
* [EMAIL PROTECTED]** http://www.crompton.com*___
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Idefisk security fix - was [Asterisk-Users] Asterisk 1.2.9 and 1.0.11 Released -- Security Fix
We released a critical update for idefisk. (Version 1.37 now ships with a patched iaxclient library). Everybody is urged to update asap. ( http://www.asteriskguru.com/idefisk/free/ ) A big thanks to coresecurity and Steve Kann for the early warning. Zoa. The Asterisk Development Team wrote: The Asterisk Development Team today released Asterisk 1.2.9 and Asterisk 1.0.11 to address a security vulnerability in the IAX2 channel driver (chan_iax2). The vulnerability affects all users with IAX2 clients that might be compromised or used by a malicious user, and can lead to denial of service attacks and random Asterisk server crashes via a relatively trivial exploit. All users are urged to upgrade as soon as they can practically do so, or ensure that they don't expose IAX2 services to the public if it is not necessary. The release files are available in the usual place (ftp.digium.com), as both tarballs and patch files relative to the last release. In addition, both the tarballs and the patch files have been signed using GPG keys of the release maintainers, so that you can ensure their authenticity. Thank you for your support of Asterisk! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Configuring Polycom 501 IP phones via the console
In my experience, this can be pretty cumbersome. I could be wrong but I think the reason I stopped doing it was that the phone would restart when you applied ANY changes, and you'd have to wait like 90 seconds or more to be able to re-access the phone via http. Moj Avi Miller wrote: Stephen Bosch wrote: All I want to do is get my Polycom 501 to register with a working Asterisk server. I want to do the configuration locally on the phone through the console. The console is very tedious. Why not use the web interface instead? Let the phone get an IP address via DHCP and then point a web browser at the phone. :) Much easier to navigate/configure. Password is the same as the advanced password on the phone itself. cYa, Avi -- Mojo [EMAIL PROTECTED] Office Manger, Horan Company, LLC (907) 747- x112 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Compiling QuaBri cards
Tzafrir Cohen a écrit : wget http://rapid.dotsrc.org/rapid/pool/main/z/zaptel/zaptel-modules-2.6.8-2-686_1.2.5-4+2.6.8-16sarge1_i386.deb It use as dependance the zaptel deb package, but in the website, only release for i386 is available, is it good?? When i depackage it, i have the errors: (Lecture de la base de données... 25603 fichiers et répertoires déjà installés.) Préparation du remplacement de zaptel 1:1.2.5-4 (en utilisant zaptel_1.2.5-4_i386.deb) ... Dépaquetage de la mise à jour de zaptel ... Paramétrage de zaptel (1.2.5-4) ... Zaptel cards initial configuration: FATAL: Error inserting ztdummy (/lib/modules/2.6.8-2-686/extra/ztdummy.ko): Unknown symbol in module, or unknown parameter (see dmesg) FATAL: Error running install command for ztdummy zaptel. Do you have any idea?? Also i am not sure that the zaptel package furnish on the site is the last one... Best regards, Olivier S. Alternatively, install zaptel-source from there and build using m-a a-i zaptel -- Olivier Saulnier STEGANUX 1er étage Diamecans Bel Air 03410 St Victor T: 04.70.02.27.62 F: 04.70.09.97.41 http://www.steganux.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] weather
I have a small Cepstral howto on my blog.. http://www.voipphreak.ca/archives/269-Even-More-Asterisk-Weather-Now-Cepstral.html On 06/06/06, David K Parker [EMAIL PROTECTED] wrote: http://nerdvittles.com/index.php?p=134 On 6/6/06, Khaled Chehab [EMAIL PROTECTED] wrote: Please any one knows how to configure the weather on asterisk or if there a weather channel I can subscribe to it * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DTMF feedthru again...
AVT??? I have ulaw allowed (only) - When you call your cell via pstn/spa-3000/* and listen on both while pressing dtmf do you hear good clean tones of enough duration to allow detection, in both directions? Do you access DTMF required services over pstn, like banking, vm, etc from local * system? Doug On Tue, 6 Jun 2006, Tom Vile wrote: Using AVT in my sipura with above settings and it work fine going out the PSTN. There was an issue a while back with an older version of Asterisk with one of my providers but it has been fine since the upgrade. I also use ulaw for calls. On 6/6/06, Doug Crompton [EMAIL PROTECTED] wrote: Tried that makes no difference. Did it for you? What DMF method(s) are you using. Looking at a goggle search yields lots of talk on this but no real solution. Apparently there is an rfc2833 issue and * is working on it??? Also it appears the codec used might be an issue. This is a serious problem in my book. It precludes me from using any DTMF over PSTN with * at this point. Any further help or explanation would be appreciated. On Tue, 6 Jun 2006, Tom Vile wrote: try setting dtmf playback length to .5 in the admin section of the Sipura and try again. On 6/6/06, Doug Crompton [EMAIL PROTECTED] wrote: Ok trying this again... is there anyone using the SPA-3000 with * I am not sure if this is a specific problem to it or not. This is something I really need to fix!!! When dialing out using * interfaced to an SPA-3000, fxs,fxo, I cannot access (reliably) DTMF menus at the called party, after call completion. Dialing DTMF is fine. I checked by calling myself. Listening to either end on a completed call, and pressing a DTMF button on the opposing phone results in an audible click and very little if any audible DTMF energy being heard. What is muting the DTMF??? Does * have anything to do with this? I am not using any dial flags. I tried 'inband' in all places with no difference. At one point this seemed like a * feature problem and I thought removing dial flags fixed it but that does not now seem to be the case. How can I fix this??? Doug * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307* * * * [EMAIL PROTECTED]* * http://www.crompton.com * ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- 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 provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Those that sacrifice essential liberty to obtain a little temporary safety deserve neither liberty nor safety. -- Ben Franklin (1759) * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307* * * * [EMAIL PROTECTED]* * http://www.crompton.com * ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- 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 provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Those that sacrifice essential liberty to obtain a little temporary safety deserve neither liberty nor safety. -- Ben Franklin (1759) * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307* * * * [EMAIL PROTECTED]* * http://www.crompton.com * ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DTMF feedthru again...
The only thing I have found that tends to point to an * problem is http://bugs.digium.com/view.php?id=6667 It is a long read and I have no ideas what the disposition is. It was a discussion back in late March. This seems to apply to all or many SIP connected devices and around implementation of the RFC. Someone had an rtp patch which they claimed worked and it later was taken back out. Digium is working on it. These are a few of the things I get from the thread. Doug On Tue, 6 Jun 2006, Mike Lynchfield wrote: in Fact we saw similar problems with all sipura products. We think its a default value thats not quite right for the north american market, these units are built and tested in asia mostly. one simple test to check it out is call this number www.nextwavetitaniumplus.com Toll-Free Account Information Line: 888-252-9535 it just seemd that even the cisco is not passing the dtmf .. Can anyone confirm ? On 6/6/06, Doug Crompton [EMAIL PROTECTED] wrote: Also to expand on this... when listening to opposing phone in a connected call over PSTN you hear a click followed by a very short burst of DTMF audible energy. Same in both directions. I can't be the only one having this problem! Doug On Tue, 6 Jun 2006, Tom Vile wrote: try setting dtmf playback length to .5 in the admin section of the Sipura and try again. On 6/6/06, Doug Crompton [EMAIL PROTECTED] wrote: Ok trying this again... is there anyone using the SPA-3000 with * I am not sure if this is a specific problem to it or not. This is something I really need to fix!!! When dialing out using * interfaced to an SPA-3000, fxs,fxo, I cannot access (reliably) DTMF menus at the called party, after call completion. Dialing DTMF is fine. I checked by calling myself. Listening to either end on a completed call, and pressing a DTMF button on the opposing phone results in an audible click and very little if any audible DTMF energy being heard. What is muting the DTMF??? Does * have anything to do with this? I am not using any dial flags. I tried 'inband' in all places with no difference. At one point this seemed like a * feature problem and I thought removing dial flags fixed it but that does not now seem to be the case. How can I fix this??? Doug * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307* * * * [EMAIL PROTECTED]* * http://www.crompton.com * ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- 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 provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Those that sacrifice essential liberty to obtain a little temporary safety deserve neither liberty nor safety. -- Ben Franklin (1759) * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307* * * * [EMAIL PROTECTED]* * http://www.crompton.com * ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Those that sacrifice essential liberty to obtain a little temporary safety deserve neither liberty nor safety. -- Ben Franklin (1759) * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307* * * * [EMAIL PROTECTED]* * http://www.crompton.com * ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk + Linksys PAP2-NA / Call Clearing
I have a handful of Linksys PAP2-NA's all talking nicely to Asterisk using standard telephones. I've been running them for the better part of this year. No complaints whatsoever. We chose the PAP2-NA's mainly due to cost and especially the ease of provisioning. In an effort to inexpensively bridge our office PBX (InterTel Axxess) to our VoIP network, we've opted to connect PAP2-NA's to the InterTel as we met with nothing but failure trying to get the InterTel to talk directly to Asterisk (InterTel problem, not *'s - SIP on the Axxess platform is broken, and MGCP doesn't seem to be compatible with *). The problem I'm having now is with call clearing when the remote end hangs up first. I'm not sure if what I'm seeing is an Asterisk or Linksys problem. When I terminate a call locally, the call clears fine. When the remote-end hangs up the call, the PAP2-NA does nothing - no hookflash, no dialtone, nothing. Eventually, the PAP2-NA bursts out a fast-busy at you. The PAP2-NA will tell * Busy Here if another call comes at it while it is in this state. Has anyone run into this, or have suggestions on how to troubleshoot further? -- +---+ ^__^ | Shane DeRidder [EMAIL PROTECTED] | . (oo)\__ | Principle Member http://silicondairy.net/ | o (__)\ )\/\ | Silicon Dairy, LLC. 802.846.4433 x101| 0||---w | +---+ |||| signature.asc Description: OpenPGP digital signature ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
