[Asterisk-Users] How to use Use different ports to authenticate SIP/IAX users
Is there a way to config a sip user so that he appears to be connecting from a different IP address? I want to use different IP addresses to authenticate different accounts with service providers rather than the username/password combo. Are there SIP settings to allow that? /Obelix This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Ask for config files of Nortell Meridian Op11 Asterisk for PRI
Hi, We have done it a little different...that said though, we have had periods of weird things happening, like digits dropping, (we blame the Nortel though!, so not the gospel ; ) : zaptata.conf [channels] context=incoming switchtype=euroisdn pridialplan=local signalling=pri_net usecallerid=yes hidecallerid=no callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes echocancel=yes rxgain=0.0 txgain=0.0 immediate=no callprogress=no callerid=asreceived group=1 callgroup=1 pickupgroup=1 channel = 1-15,17-31 zaptel.conf loadzone = za defaultzone=za span=1,1,0,ccs,hdb3,crc4 bchan=1-15 dchan=16 bchan=17-31 Anthony Rodgers wrote: Wow - I thought we were the only ones doing this. OK - here goes. Our zaptel.conf looks like this: span=1,1,0,esf,b8zs bchan=1-23 dchan=24 #clear=1-24 loadzone = us defaultzone=us Our zapata.conf looks like this: [trunkgroups] [channels] context=incoming switchtype=5ess usecallingpres=yes echocancel=128 usecallerid=yes echocancelwhenbridged=yes echotraining=yes echotraining=800 rxgain=-4.0 txgain=-6.0 group=1 callgroup=1 pickupgroup=1 signalling = pri_net channel = 1-23 musiconhold=default OK. that's the easy part :-) On the Option 11C, there is a 9-page PDF on how to do it - I can send it direct if you send me your email addresses, to avoid sending it to the entire list. The only issue we face with the way we have it set up is that the Nortel won't send name and number for caller ID, only number - if we figure that out, I'll update the list. Of course, if anyone else already knows :-) Regards, ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- MICHAEL TOOP Tel 011 602 9300 Fax 011 656 1342 Mobile 083 364 2370 Web www.bizcall.co.za ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cannot telnet to port 5038 on asterisk
- Original Message - From: Tzafrir Cohen [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Monday, October 17, 2005 7:18 PM Subject: Re: [Asterisk-Users] Cannot telnet to port 5038 on asterisk On Mon, Oct 17, 2005 at 06:56:39PM +0100, Steve Daniels wrote: Try a a good old netstat -a | grep 5038 netstat -lntup | grep 5038 -l is generally preferable than -a: only gives listening ports. -t: tcp sockets only. -p: also pid and name of the process that has the socket. -a show's all sockets regardless of state though I admit including -p would check that it was actually * listening on that port, and make sure that no other process is preventing * from binding to that port. That will tell you if * is listening and what it's listening on. Then if it show's * is listening, it must be a permit =, or a firewall issue. Though most firewalls allow everything on the interface lo. My guess is that either asterisk is not running or it has not been *restarted* since the configuration change. Reloading is not enough to change the IP address(es) Asterisk listens on or make it start listen at all (right?) Make sure you restart the process, it sounds as if you've probably restarted it a fair few times though.. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Middle Ground between POTS and T1?
Here (Croatia) is also possible to get partial E1 (PRI/PRA)... from one of our telcos (DT T-com) we can get PRA in 10 increments: 10B, 20B and 30B We have a partial T1 (5B + D, iirc) from Allstream - there may be a provider in your area that does something similar. Regards, -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web: http://www.dnv.org RSS Feed: http://www.dnv.org/rss.asp On 17-Oct-05, at 12:48 PM, Matthew T. O'Connor wrote: I was wondering if there was a middle ground between POTS lines and a T1. I have a new office with a T1 line and while it's working well, it's a lot of money and we will never use anywhere near 23 lines at one time. Is it possible to get a few ISDN lines or something and bundle them together? Basically I would like to get the digital features of the T1 PRI (DID number, etc...) but smaller. Thanks, Matthew ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Teliax IAX problems -- Asterisk doesn't see answer
I think I ran in to this problem a while back as well. I'm also running a CVS version of Asterisk. I talked to David and he switched me to SIP from their gateway to their Asterisk proxy which solved the issue. On Mon, 2005-10-17 at 22:05 -0500, Rob Fugina wrote: On 10/17/05, Rich Adamson [EMAIL PROTECTED] wrote: I've been trying to diagnose the same problem with teliax, and it seems to be a jitterbuffer problem. Since turning it off, we've not had a problem. My guess is that teliax servers are not current code, or they've modified the code for some reason. The tech that I corresponded with suggested 'gsm' was the culprit, but I had the same issues with g729, ilbc, etc. Try jitterbuffer=no in the appropriate section of your iax.conf and restart asterisk. Well, that did it. Here's hoping that teliax sees the light soon... If having no jitter buffer becomes a problem, I won't hesitate to route my calls through somebody else... Thanks for the help1 Rob ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mike Benoit [EMAIL PROTECTED] signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Talkoff (Spurious DTMF) with 1.0.9.2 and TE406P
We recently migrated a couple of PRIs to Asterisk 1.0.9.2 and a TE406P and are getting reports of talkoff (spurious/random DTMF tones heard by people on SIP equipment connected to the Asterisk server. We previously were using 1.0.3 with a T100P without any talkoff. (a) We have not set relaxdtmf. (b) There is no apparent pattern to the problem (seems to affect users on ATAs as well as SIP phones from different manufacturers). Any suggestions? -- George Pajari, netVOICE communications604 484 VOIP (484 8647 x102) Open Source VoIP/Telephony Specialists 1 877 NET VOIP (638 8647 x102) www.netvoice.ca www.ip-centrex.ca www.digium.ca www.grandstream.ca www.sipura.ca www.snom.ca ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] free dids on goiax.com
GoIAX, the Asterisk community's free IAX provider, is offering free US dids now. I loaded about 175 dids in and put up a very beta sign in page. Unfortunately I had to restrict the free us/canada outbound calling back down to toll-free only. There was a lot of war dialing and prank calling going on. I'm working on some stuff to hopefully curb that kind of stuff down so I can unrestrict outdial again, but this is the problem with free.. there is always someone that will abuse it. If anybody has any ideas on how to keep the abuse down let me know. The best ideas I have now is to only allow a certain amount of calling per month, add velocity checking, and somehow put some accountability into the sign up process to keep the prank callers and multiple account abusers away. yours, Matthew Simpson GoIAX -- www.goiax.com TxLink -- www.txlink.net ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] huge problem compiling * on gcc4.x (SUSE 10.0)
On Mon, Oct 17, 2005 at 07:01:17PM -0400, Walt Reed wrote: On Sun, Oct 16, 2005 at 09:21:09PM +0200, [Ludwig IT-Services - GMAIL ] - Michael Ludwig said: I'm very new to this list and to asterisk and stuff at all. To build my asterisk server I installed a new machine running the new SUSE Linux 10.0 (retail version on DVD). I need asterisk (tried 1.0.9), bristuff (off junghanns.net, -0.2.0-RC8o) and the florz-patch because I have two HFC-S-ISDN cards in that machine. Now when it comes to compiling I get a huge bunch of warnings and stuff, zaptel 1.0.9.2 fails to compile and asterisk 1.0.9 also fails to compile. SUSE 10.0 uses gcc 4.0.2 and as I asked in some other mailing list and forums, that is the reason why * stuff fails to compile. Is there any stable asterisk version available which does compile fine on a gcc4.x ? If not, will the * source be changed to finely compile on gcc 4.x? If yes, when will that be? (I need the * stuff now). If not, why not? What's on with the 1.2.0-beta stuff out there on the asterisk.org webpages? Does that one compile on gcc4.x ? I've been running a * (cvs HEAD) instance on Debian unstable, which has upgraded to gcc 4. Gcc 4 still has problems compiling the kernel (as of 2.6.12) on debian, and you want to use the same version of the compiler on the zaptel modules that you do on the kernel. I was unable to get a clean compile of the kernel or * with gcc 4. You can ask about this in Debian lists. I don't have unstable so I can't test for myself, but the current unstable kernel surely builds for them. As for Asterisk 1.2: It should hit experimental any day now. There are also unofficial debs at http://rapid.dotsrc.org/experimental/ . If those don't build with gcc 4 then this should be reported. Most gcc 4 incompatibility bugs I saw were fixed pretty fast. Asterisk 1.0.9 and bristuff RC8l (the current package in unstable) probably builds with gcc4. I'm not sure if it has the florz patch in it. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CAPI - displaying individual MSN
Hi,
I'm currently using chan_capi-cm-0.6, with the following capi.conf:
[general]
nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8
language=de
[ISDN1]
msn=8304490
incomingmsn=8304490
isdnmode=msn
group=1
controller=1
softdtmf=1
context=demo
echosquelch=1
echocancel=yes
echotail=64
callgroup=1
devices=2
Each user has a different numer, e.g. 83044910, 83044911, 83044912 and
so on.
This number should appear on the display of the called party, but how do
I configure this?
With the above configuration the display always shows 8304490.
I've tried to change the number in the dialplan, but this doesn't change
anything:
exten = _90[23456789].,1,SetCIDNum(83044912)
exten = _90[23456789].,2,Dial(Capi/g1/${EXTEN:1},30,tr)
If I remove the mns line in the capi.conf or set msn=* or msn=830449*
Asterisk isn't able to open the CAPI channel.
Does anyone have a hint for me?
If yes - THANK YOU ;-)
Stefan
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RE: [Asterisk-Users] free dids on goiax.com
That really is a shame, goiax.com has been the best free termination service I have seen. The call quality was excellent, better then some paid services I have used. One idea, I'm not sure if you already did it, only allow one concurrent call per account? And now DIDs, thanks from all of us for the great service. Kevin -Original Message- From: Matthew Simpson [mailto:[EMAIL PROTECTED] Sent: October 18, 2005 2:05 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] free dids on goiax.com GoIAX, the Asterisk community's free IAX provider, is offering free US dids now. I loaded about 175 dids in and put up a very beta sign in page. Unfortunately I had to restrict the free us/canada outbound calling back down to toll-free only. There was a lot of war dialing and prank calling going on. I'm working on some stuff to hopefully curb that kind of stuff down so I can unrestrict outdial again, but this is the problem with free.. there is always someone that will abuse it. If anybody has any ideas on how to keep the abuse down let me know. The best ideas I have now is to only allow a certain amount of calling per month, add velocity checking, and somehow put some accountability into the sign up process to keep the prank callers and multiple account abusers away. yours, Matthew Simpson GoIAX -- www.goiax.com TxLink -- www.txlink.net ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Pb musiconhold with G729 codec
Hi, When i place a call on hold, and then return to it, the caller then hears my voice in a delay usually equal to the amount of time i put them on hold. I have the problem only with G729 codec and with my voip provider (i live in france and i use wengo) My configuration : - Pentium III 550 Mhz + 256 Mo Ram - [EMAIL PROTECTED] 1.5 - Grandstream 102 IP Phone - TDM400p card (2FXO + 1 FXS) - 3 licences G729 Codec ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] free dids on goiax.com
On Tue, 2005-10-18 at 02:36 -0500, Kevin Scott wrote: That really is a shame, goiax.com has been the best free termination service I have seen. The call quality was excellent, better then some paid services I have used. One idea, I'm not sure if you already did it, only allow one concurrent call per account? And now DIDs, thanks from all of us for the great service. Kevin That solves only part of the problem and is easily worked around. One problem is mass calls, either for wardialing or call centers doing telemarketing. And does nothing for prank calls, but you can almost never stop those. Tracking IPs in registration can help weed out some (but not all) mlutiple account users. That also makes it hard for roommates to each have their own accounts as they would most likely be using NAT. This coupled with a group count of 1 per account can help mitigate but not eliminate war dialers/telemarketers from abusing the service. The reality is that it will most likely take multiple tactics ... A credit system per account can be implemented, where duty cycle determines when the next call can be placed (ie avoid continous calling out by forcing them to not have concurrent calls for a while, if they stay off for a while they can build credits and make a few calls in rapid succession before being turned down). The shorter the duration each call is potentially the greater the chance they are doing something undesirable. Telemarketers and war dialers tend to not stay on the call for 20 minutes or more ... Another method is to just put a cap not more than X calls per Y timeframe can be placed. That will slow but not prevent it. Take for example a group of people who do a distributed war dialing project. If they all have 10 calls per hour and there are more than 6 of them then continous calls can be placed, providing the calls are 1 minute in duration. For authentication you can take the ebay approach. Ban free mail providers. Of course getting the list of free mail providers is the trick. This mitigates but does not eliminate people using multiple accounts. I have a couple domains myself, friends also have several domains. I could in theory have 100 email addresses all completly different, and all 'non-free'. Checking MX records to see where the mail goes to see if its all going to the same machine might help but adds a ton of overhead to the process. Checking IPs on signup wouldnt be that effective given that there are thousands of proxy servers all over, making that almost impossible to prevent. It would just add another hurdle for someone to leap over, the more there are the more likely people will not bother becuase it wont be worth it, but there are always those dedicated few. Programs like proxychains aide in even using an iax client to connect via proxies, providing the proxy supports all the required protocols. I dont think you can stop it, only make it hard to use for the undesirable purposes, at the risk of making it so hard to use that no one will want to use it, which generally is a bad thing. I am also sure that there are other things that someone else can contribute that used in combination or in stead of my suggestions can make this harder to wardial/telemarket through but easy enough for everyone else to use. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] free dids on goiax.com
On 18 Oct 2005, at 08:05, Matthew Simpson wrote: GoIAX, the Asterisk community's free IAX provider, is offering free US dids now. I loaded about 175 dids in and put up a very beta sign in page. Fantastic, got one, thanks. Unfortunately I had to restrict the free us/canada outbound calling back down to toll-free only. There was a lot of war dialing and prank calling going on. I'm working on some stuff to hopefully curb that kind of stuff down so I can unrestrict outdial again, but this is the problem with free.. there is always someone that will abuse it. That's a shame, it is a great service but, as you say, inevitable that some would take advantage. If anybody has any ideas on how to keep the abuse down let me know. The best ideas I have now is to only allow a certain amount of calling per month, add velocity checking, and somehow put some accountability into the sign up process to keep the prank callers and multiple account abusers away. You could restrict it to folks with DIDs and make sure their DID goes out on outbound calls, giving the victims of pranks a place to complain. You could also change the signup so that we have to provide a pots number as a contact point. Otherwise I'd be happy to be limited to (say) 100 different numbers I can call, that would limit the wardialers at least. Keep up the good work. Tim. yours, Matthew Simpson GoIAX -- www.goiax.com TxLink -- www.txlink.net ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Slow dialling from PBX into * via E1
Hi :) I have a little 'slow dialling' problem. When I dial, e.g. 200# for the Asterisk 'echo test' demo application from my PBX extension 1010, I see this in the console the instant I press the # key: -- Starting simple switch on 'Zap/65-1' -- Accepting overlap call from '1010' to '200' on channel 0/3, span 3 then exactly 3 seconds elapses, and finally -- Executing Playback(Zap/65-1, demo-echotest) in new stack -- Playing 'demo-echotest' (language 'en') at which point Allison's 'sultry' voice announces 'You are about to enter an echo test...' Can I remove this 3 second pause? It's rather annoying, and it doesn't happen when I dial out from the PBX to a normal PSTN line or to a PBX-provided extension, or indeed to any local ISDN2 device. Even with debug + verbose both at 99, I see no extra information I'm using a Sangoma A104u with wanpipe-beta15-2.3.3.tgz and Asterisk/Zaptel/Libpri 1.2.0-beta1. The extensions.conf is trivial [general] static=yes writeprotect=yes [fromaxxess] exten = 200,1,Playback(demo-echotest) ; Let them know what's going on exten = 200,2,Echo ; Do the echo test exten = 200,3,Playback(demo-echodone) ; Let them know it's over Sangoma's support can't understand how it can be their drivers / cards causing the issue since there is no buffering at all in Zaptel (and let's face it that makes sense :) Cheers, Gavin. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] fax device behind TDM400P
Hello, iam trying to connect an analogue Fax (as opposed to a ISDN Fax device) behind a TDM400P. However, when i connect the Fax to the Card, asterisk shows it as always being offhook. Iam currently out of ideas what might be wrong. The Fax device is connected using a 1:1 four-wire RJ cable. The setup in zapata.conf is: | callwaiting=0 | context=fax_out | faxdetect=both | adsi=0 | callerid=309 | channel =2 -- http://www.ukeer.de/about.html ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] error while writing audio data: : Broken pipe
Dear Asterisk developers, I run the same asterisk version on the home machine and on the work. On the home machine I have Slackware 10.0 (kernel 2.4.24) while on the work machine I have Mandrake 10.1 (kernel 2.6.8.1). When I run asterisk on the work machine, these warnings and error appear (there are no warnings or error at home): [ Booting..Oct 17 18:19:04 WARNING[9036]: res_musiconhold.c:580 moh_register: Unable to open pseudo channel for timing... Sound may be choppy. Warning, flexibel rate not heavily tested! ...Oct 17 18:19:04 WARNING[9036]: chan_iax2.c:7477 load_module: Unable to open IAX timing interface: No such file or directory ..Oct 17 18:19:04 WARNING[9036]: chan_skinny.c:2587 reload_config: Unable to get our IP address, Skinny disabled Oct 17 18:19:04 WARNING[9036]: loader.c:258 ast_load_resource: /usr/lib/asterisk/modules/chan_features.so: undefined symbol: ast_register_file_version Oct 17 18:19:04 WARNING[9036]: loader.c:440 load_modules: Loading module chan_features.so failed! [EMAIL PROTECTED] rc.d]# Ouch ... error while writing audio data: : Broken pipe All the Asterisk configuration files on both machines the same, again. What thing may cause this? Much thanks. Best regards, Corrado Mastruzzi ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CAPI - displaying individual MSN
Stefan Günther schrieb:
With the above configuration the display always shows 8304490.
I've tried to change the number in the dialplan, but this doesn't change
anything:
exten = _90[23456789].,1,SetCIDNum(83044912)
Try to use SetCallerID instead of SetCIDNum and see if it helps.
exten = _90[23456789].,2,Dial(Capi/g1/${EXTEN:1},30,tr)
--
Best regards
Peer Oliver Schmidt
PGP Key ID: 0x83E1C2EA
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Re: [Asterisk-Users] CAPI - displaying individual MSN
Hi!
msn=8304490
incomingmsn=8304490
Each user has a different numer, e.g. 83044910, 83044911, 83044912 and
so on.
This number should appear on the display of the called party, but how do
I configure this?
With the above configuration the display always shows 8304490.
I've tried to change the number in the dialplan, but this doesn't change
anything:
exten = _90[23456789].,1,SetCIDNum(83044912)
exten = _90[23456789].,2,Dial(Capi/g1/${EXTEN:1},30,tr)
If I remove the mns line in the capi.conf or set msn=* or msn=830449*
Asterisk isn't able to open the CAPI channel.
You need to modify incomingmsn= and not msn= for this to work as
expected. Also be aware that often these two settings require different
values for the same meaning, e.g. you might have to add the area prefix
for the msn= setting (40 for Hamburg, 89 for München etc). If however
your Asterisk is behind a PBX then your incoming MSN might only have to be 910,
911 and 912.
The above applies also to your SetCIDNum statement, it must match a
valid (!) MSN.
Cheers, Philipp
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Re: [Asterisk-Users] CAPI - displaying individual MSN
On Tue, 18 Oct 2005, Stefan Günther wrote:
..
Each user has a different numer, e.g. 83044910, 83044911, 83044912 and
so on.
This number should appear on the display of the called party, but how do
I configure this?
With the above configuration the display always shows 8304490.
I've tried to change the number in the dialplan, but this doesn't change
anything:
exten = _90[23456789].,1,SetCIDNum(83044912)
exten = _90[23456789].,2,Dial(Capi/g1/${EXTEN:1},30,tr)
If I remove the mns line in the capi.conf or set msn=* or msn=830449*
Asterisk isn't able to open the CAPI channel.
msn= does not exist anymore, it has no effect.
Use incomingmsn=* to specify which MSN shall be handled by Astreisk.
Are you sure you have PtMP (MSN) connection? When you have numbers like
83044910, 83044911, 83044912,... and the display shows 8304490, then it
looks like a PtP connection with base number 830449-X.
If thats the case, you should
- switch to isdnmode=did
- SetCIDNum(12), instead of SetCIDNum(83044912)
Armin
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[Asterisk-Users] Custom Callback
Hello All, I am trying to create a custom callback on asterisk. [custom-callback] exten = s,1,Wait(2) exten = s,2,Hangup exten = s,3,Dial(Zap/g0/92962676) exten = s,4,DigitTimeout(5) exten = s,5,ResponseTimeout(10) exten = s,6,Authenticate(1234) exten = s,7,DISA(no-password|from-internal) exten = s,8,Hangup If I ring my DID and if Caller ID is my cell number, the asterisk server should hang up the call, Issue a Congestion to my phone and call me back on a set cell phone number(doesn't need to be the same as the CID). How do I do this on this thing? I dunno much about AGI so any pointers would be nice:) Thanks, Dinesh. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] error while writing audio data: : Broken pipe
Corrado wrote: Dear Asterisk developers, I run the same asterisk version on the home machine and on the work. On the home machine I have Slackware 10.0 (kernel 2.4.24) while on the work machine I have Mandrake 10.1 (kernel 2.6.8.1 http://2.6.8.1). When I run asterisk on the work machine, these warnings and error appear (there are no warnings or error at home): [ Booting..Oct 17 18:19:04 WARNING[9036]: res_musiconhold.c:580 moh_register: Unable to open pseudo channel for timing... Sound may be choppy. Warning, flexibel rate not heavily tested! ...Oct 17 18:19:04 WARNING[9036]: chan_iax2.c:7477 load_module: Unable to open IAX timing interface: No such file or directory ..Oct 17 18:19:04 WARNING[9036]: chan_skinny.c:2587 reload_config: Unable to get our IP address, Skinny disabled Oct 17 18:19:04 WARNING[9036]: loader.c:258 ast_load_resource: /usr/lib/asterisk/modules/chan_features.so: undefined symbol: ast_register_file_version Oct 17 18:19:04 WARNING[9036]: loader.c:440 load_modules: Loading module chan_features.so failed! [EMAIL PROTECTED] rc.d]# Ouch ... error while writing audio data: : Broken pipe All the Asterisk configuration files on both machines the same, again. What thing may cause this? Much thanks. Best regards, Corrado Mastruzzi ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I've seen this error message before after upgrading kernel # Ouch ... error while writing audio data: : Broken pipe On the work machine do you have any digium hardware if so have you run rebuild_zaptel and genzaptelconf? IIRC WARNING[9036]: res_musiconhold.c:580 moh_register: Unable to open pseudo channel for timing... Sound may be choppy. with a 2.6 kernel you will get these errors as there is nothing to provide the relevant timing events. Bails ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Recomendations for utility to generate Asterisk configuration
I need some help generating configuration files for Asterisk. Since I'm running under Solaris I'm having trouble with some of the utilities that are more linux-centric. Can anyone recommend a free/low-cost package to generate conf files that is not linux-dependent and will handle a IAX2 and SIP trunk? Frank ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Queues and call waiting indication
Hi, I'm running 1.2 beta1 in a mini call center. I have 3 queues with 10 operators, and I'm running into some trouble because when all the operators are busy answering call asterisk still sends them more, resulting in a beep beep (call waiting) over and over again in Xlite audio. An easy solution woud be the use of a single line user agent, like firefly, still this behaviour does not make any sense to me. I tried using incominglimit and outgoinglimit in my sip.conf, even if they are deprecated: no luck. Here is a sample from my queues.conf, something wrong in my setup maybe ? Tnx for any help! [ingombranti] joinempty = strict maxlen=3 musiconhold = default announce = annuncio-ingombranti strategy = rrmemory servicelevel = 60 timeout = 15 announce-frequency = 15 eventwhencalled = yes member=SIP/401 member=SIP/402 member=SIP/403 member=SIP/404 member=SIP/405 member=SIP/406 member=SIP/407 member=SIP/408 member=SIP/409 member=SIP/410 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Agent recording and muxmon
I was wanting to use the new MuxMon application to record calls - it seems to be a nicer way of recording than the Monitor application. However, there is a slight issue with agents - we use the recordcalls option in agents.conf to record inbound agent calls - and I believe from looking at the source code that is uses the monitor application. Is there any way to get chan_agent to use muxmon instead of monitor, or a) Do I have to patch chan_agent.c b) Can I modify my dialplan to use muxmon and remove the record calls option from agents.conf ? Julian. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 411
carl is the link. http://www.tmcnet.com/usubmit/2005/Aug/1170660.htm___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
SV: [Asterisk-Users] Queues and call waiting indication
Hi, This issue has been discussed probably a million times on every asterisk forum in the world and I have the same problem too. Another problem you would have with the agents is that when they make an outgoing call they are not regarded as busy by asterisk and it sends more calls to the agent if it has call waiting enabled. This behaviour is totally senseless since the whole purouse of queues is to _queue_ the callers until the agent is available. available usually means not on the phone -- whether or not it's an incoming or outgoing call. I solved this problem by using single-line clients and phones where you can turn off call wating. //Jan -Ursprungligt meddelande- Från: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] För [EMAIL PROTECTED] Skickat: den 18 oktober 2005 14:14 Till: asterisk-users@lists.digium.com Ämne: [Asterisk-Users] Queues and call waiting indication Hi, I'm running 1.2 beta1 in a mini call center. I have 3 queues with 10 operators, and I'm running into some trouble because when all the operators are busy answering call asterisk still sends them more, resulting in a beep beep (call waiting) over and over again in Xlite audio. An easy solution woud be the use of a single line user agent, like firefly, still this behaviour does not make any sense to me. I tried using incominglimit and outgoinglimit in my sip.conf, even if they are deprecated: no luck. Here is a sample from my queues.conf, something wrong in my setup maybe ? Tnx for any help! [ingombranti] joinempty = strict maxlen=3 musiconhold = default announce = annuncio-ingombranti strategy = rrmemory servicelevel = 60 timeout = 15 announce-frequency = 15 eventwhencalled = yes member=SIP/401 member=SIP/402 member=SIP/403 member=SIP/404 member=SIP/405 member=SIP/406 member=SIP/407 member=SIP/408 member=SIP/409 member=SIP/410 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Agent recording and muxmon
If you're using AgentCallBackLogin it should be fairly easy to to do what you're looking for in step 'b'. On 10/18/05, Julian Lyndon-Smith [EMAIL PROTECTED] wrote:I was wanting to use the new MuxMon application to record calls - it seems to be a nicer way of recording than the Monitor application.However, there is a slight issue with agents - we use the recordcallsoption in agents.conf to record inbound agent calls - and I believe from looking at the source code that is uses the monitor application.Is there any way to get chan_agent to use muxmon instead of monitor, ora) Do I have to patch chan_agent.cb) Can I modify my dialplan to use muxmon and remove the record calls option from agents.conf ?Julian.___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing list Asterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] fax device behind TDM400P
Hello, iam trying to connect an analogue Fax (as opposed to a ISDN Fax device) behind a TDM400P. However, when i connect the Fax to the Card, asterisk shows it as always being offhook. Iam currently out of ideas what might be wrong. The Fax device is connected using a 1:1 four-wire RJ cable. The setup in zapata.conf is: | callwaiting=0 | context=fax_out | faxdetect=both | adsi=0 | callerid=309 | channel =2 Rico, I assume you are connecting the fax to an FXS port on your TDM400P as FXO will not work. Have you tired a different RJ11 cable between the FXS port and the fax. Surprisingly, cables seem to be one of the most common causes of these types of problems. It the cable was made poorly or stretched too much at some point, there may be a short which will cause the offhook condition. Finally, connect a standard analog telephone to that port and see if you have any issues with offhook and dialtone. If all is well with the ananlog phone, then you can safely assume that the problem is your fax machine. If you have a spare fax, give it a try. If you are in a business area, maybe you can borrow a fax from a neighbor. Finally, if you cannot locate a fax to test, here is a cheap ($30 after rebate) fax that may meet your needs for testing and actually may make a great backup fax. http://shop4.outpost.com/product/3353422?site=sr:SEARCH:MAIN_RSLT_PG Thanks, Steve Totaro ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: SV: [Asterisk-Users] Queues and call waiting indication
This behaviour is totally senseless since the whole purouse of queues is to _queue_ the callers until the agent is available. available usually means not on the phone -- whether or not it's an incoming or outgoing call. Agree! I solved this problem by using single-line clients and phones where you can turn off call wating. Can you suggest me a SIP or IAX phone with just one line that can also open url's passed by asterisk ? Tnx! //Jan -Ursprungligt meddelande- Från: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] För [EMAIL PROTECTED] Skickat: den 18 oktober 2005 14:14 Till: asterisk-users@lists.digium.com Ämne: [Asterisk-Users] Queues and call waiting indication Hi, I'm running 1.2 beta1 in a mini call center. I have 3 queues with 10 operators, and I'm running into some trouble because when all the operators are busy answering call asterisk still sends them more, resulting in a beep beep (call waiting) over and over again in Xlite audio. An easy solution woud be the use of a single line user agent, like firefly, still this behaviour does not make any sense to me. I tried using incominglimit and outgoinglimit in my sip.conf, even if they are deprecated: no luck. Here is a sample from my queues.conf, something wrong in my setup maybe ? Tnx for any help! [ingombranti] joinempty = strict maxlen=3 musiconhold = default announce = annuncio-ingombranti strategy = rrmemory servicelevel = 60 timeout = 15 announce-frequency = 15 eventwhencalled = yes member=SIP/401 member=SIP/402 member=SIP/403 member=SIP/404 member=SIP/405 member=SIP/406 member=SIP/407 member=SIP/408 member=SIP/409 member=SIP/410 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
SV: SV: [Asterisk-Users] Queues and call waiting indication
My suggestion would be the one-line eyeBeam phone under development. Check out support.xten.com. //Jan -Ursprungligt meddelande- Från: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] För [EMAIL PROTECTED] Skickat: den 18 oktober 2005 14:48 Till: Asterisk Users Mailing List - Non-Commercial Discussion Ämne: Re: SV: [Asterisk-Users] Queues and call waiting indication This behaviour is totally senseless since the whole purouse of queues is to _queue_ the callers until the agent is available. available usually means not on the phone -- whether or not it's an incoming or outgoing call. Agree! I solved this problem by using single-line clients and phones where you can turn off call wating. Can you suggest me a SIP or IAX phone with just one line that can also open url's passed by asterisk ? Tnx! //Jan -Ursprungligt meddelande- Från: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] För [EMAIL PROTECTED] Skickat: den 18 oktober 2005 14:14 Till: asterisk-users@lists.digium.com Ämne: [Asterisk-Users] Queues and call waiting indication Hi, I'm running 1.2 beta1 in a mini call center. I have 3 queues with 10 operators, and I'm running into some trouble because when all the operators are busy answering call asterisk still sends them more, resulting in a beep beep (call waiting) over and over again in Xlite audio. An easy solution woud be the use of a single line user agent, like firefly, still this behaviour does not make any sense to me. I tried using incominglimit and outgoinglimit in my sip.conf, even if they are deprecated: no luck. Here is a sample from my queues.conf, something wrong in my setup maybe ? Tnx for any help! [ingombranti] joinempty = strict maxlen=3 musiconhold = default announce = annuncio-ingombranti strategy = rrmemory servicelevel = 60 timeout = 15 announce-frequency = 15 eventwhencalled = yes member=SIP/401 member=SIP/402 member=SIP/403 member=SIP/404 member=SIP/405 member=SIP/406 member=SIP/407 member=SIP/408 member=SIP/409 member=SIP/410 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Recomendations for utility to generate Asteriskconfiguration
I need some help generating configuration files for Asterisk. Since I'm running under Solaris I'm having trouble with some of the utilities that are more linux-centric. Can anyone recommend a free/low-cost package to generate conf files that is not linux-dependent and will handle a IAX2 and SIP trunk? Frank Frank, I am not a Solaris guy so let me offer a work around that should work well for you. If you have an old laptop or desktop that is not being used, install [EMAIL PROTECTED] on it. Use the AMP piece of [EMAIL PROTECTED] to do all of your configurations. When done, click on the red bar which writes the .conf files from the MySQL database, then use WinSCP to copy all of your confs to the Solaris box (assuming WinSCP works with Solaris). You could also just FTP the files if you have to. I think this might be the easiest way to create a working system with the benefit of the additional dialplan functionality that comes with [EMAIL PROTECTED] Thanks, Steve ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Recomendations for utility to generate Asteriskconfiguration
On Wed, Oct 19, 2005 at 09:05:45AM -0400, asterisk wrote: I need some help generating configuration files for Asterisk. Since I'm running under Solaris I'm having trouble with some of the utilities that are more linux-centric. Can anyone recommend a free/low-cost package to generate conf files that is not linux-dependent and will handle a IAX2 and SIP trunk? Frank Frank, I am not a Solaris guy so let me offer a work around that should work well for you. If you have an old laptop or desktop that is not being used, install [EMAIL PROTECTED] on it. Use the AMP piece of [EMAIL PROTECTED] to do all of your configurations. When done, click on the red bar which writes the .conf files from the MySQL database, then use WinSCP to copy all of your confs to the Solaris box (assuming WinSCP works with Solaris). You could also just FTP the files if you have to. I think this might be the easiest way to create a working system with the benefit of the additional dialplan functionality that comes with [EMAIL PROTECTED] AMP's dialplan and setup is quite complex. Requires, e.g, a number of AGIs. This is normally not the type of thing you'd like to hand-edit later after the initial adaptation to the target system. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Hints on hardware to use
Hello, I have to deploy an Asterisk PBX with this requirements: - 1 or 2 ISDN lines in input/output - 14 internal analog phones (yes, I know, analog ones... ;( ) - Billing interface for the operator (for usage of analog phones) For the external interface I'm thinking about Beronet Quad Span ISDN/BRI Card TE/NT (BN4S0). I'm in trouble about the internal interfaces: the first thought was about Digium card (4 x TDM400P with 2 x S100M modules each), but it's difficult to find a MB with 5 PCI, and I'll have no chanches for future expansions. Does anyone of you know a PCI card with 8 FXS port that SURELY works with Asterisk? I'm ready to examine any other piece of hardware with 8 or more FXS ports, too... By the way, for billing operations I'm going to check AstBill sofware; did anyone positively try it with asterisk in operational environment? Any hint will be greatly appreciated... ;) Thanks Jonathan ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] setting a dialplan on a GXP-2000 Grandstream
Hi, I looked at the docs and probably missed it: is there a way to set a dialplan on the GXP-2000? (to avoid having to press Send) Thanks, -- Computers are useless. They can only give answers. - Pablo Picasso ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Recomendations for utility to generateAsteriskconfiguration
On Wed, Oct 19, 2005 at 09:05:45AM -0400, asterisk wrote: I need some help generating configuration files for Asterisk. Since I'm running under Solaris I'm having trouble with some of the utilities that are more linux-centric. Can anyone recommend a free/low-cost package to generate conf files that is not linux-dependent and will handle a IAX2 and SIP trunk? Frank Frank, I am not a Solaris guy so let me offer a work around that should work well for you. If you have an old laptop or desktop that is not being used, install [EMAIL PROTECTED] on it. Use the AMP piece of [EMAIL PROTECTED] to do all of your configurations. When done, click on the red bar which writes the .conf files from the MySQL database, then use WinSCP to copy all of your confs to the Solaris box (assuming WinSCP works with Solaris). You could also just FTP the files if you have to. I think this might be the easiest way to create a working system with the benefit of the additional dialplan functionality that comes with [EMAIL PROTECTED] AMP's dialplan and setup is quite complex. Requires, e.g, a number of AGIs. This is normally not the type of thing you'd like to hand-edit later after the initial adaptation to the target system. Who said anything about hand editing? That is why you would want to keep the old computer running [EMAIL PROTECTED] Instead of hand editing anything, make the changes on the [EMAIL PROTECTED] box's AMP GUI and copy them over again. Very simple and most tech folks have an old computer laying around somewhere that could be put to use. Thanks, Steve Totaro ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hints on hardware to use
Hello, I have to deploy an Asterisk PBX with this requirements: - 1 or 2 ISDN lines in input/output - 14 internal analog phones (yes, I know, analog ones... ;( ) - Billing interface for the operator (for usage of analog phones) For the external interface I'm thinking about Beronet Quad Span ISDN/BRI Card TE/NT (BN4S0). I'm in trouble about the internal interfaces: the first thought was about Digium card (4 x TDM400P with 2 x S100M modules each), but it's difficult to find a MB with 5 PCI, and I'll have no chanches for future expansions. Does anyone of you know a PCI card with 8 FXS port that SURELY works with Asterisk? I'm ready to examine any other piece of hardware with 8 or more FXS ports, too... By the way, for billing operations I'm going to check AstBill sofware; did anyone positively try it with asterisk in operational environment? Any hint will be greatly appreciated... ;) Thanks Jonathan For your internal analog extensions why not get a Digium T1 card and a channel bank. I only have experience with Adtran 600E but they work extremely well and can be had used on ebay for about $600 regularly (if you are lucky you may be able to get it much cheaper.) I read the Rhino channel banks are much cheaper and work well with asterisk but have no personal experience. With the Channel bank solution, you are looking at $500 for the T1 board and another $600 for the channel bank with 24 FXS ports. Its a solid solution and gives you tons of room to upgrade from 14 FXS ports to 24 by simply adding phones. Thanks, Steve Totaro ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Display number dialled
Hi Is it possible with Asterisk to tellthe called party which number was dialled by the caller? Or in place of the number dialled have a description such as 'Sales' or 'Accounts'? Ideally, I would like to show a description corresponding to the number dialled followed by CIDName. How might this be set up? Currently my extensions.conf is: exten = xx,1,LookupCIDNameexten = xx,2,Dial(SIP/xx,50)exten = xx,3,Voicemail(xx)exten = xx,4,Hangup Thanks for your help. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Sample cisco config for cisco 7206
Jerry, here are the relevant parts of my 7206 config. Some things have been changed to protect the innocent. ;) dspint DSPfarm1/0 codec med ! isdn switch-type primary-ni ! ! voice call send-alert ! voice service pots fax protocol pass-through g711ulaw ! voice service voip fax protocol pass-through g711ulaw ! voice class codec 1 codec preference 1 g711ulaw codec preference 2 g711alaw ! controller T1 2/0 framing esf linecode b8zs cablelength long 0db pri-group timeslots 1-12,24 ! interface Serial2/0:23 no ip address no logging event link-status isdn switch-type primary-ni isdn incoming-voice voice isdn supp-service name calling isdn send-alerting isdn sending-complete isdn incoming progress validate no cdp enable ! voice-port 2/0:23 ! ! dial-peer cor custom ! ! ! dial-peer voice 1 voip service session destination-pattern . voice-class codec 1 session protocol sipv2 session target sip-server dtmf-relay rtp-nte no vad ! dial-peer voice 7999 pots service session destination-pattern ...$ port 2/0:23 ! dial-peer voice 5160 voip huntstop service session destination-pattern 51600[0-5,9]$ voice-class codec 1 session protocol sipv2 session target ipv4:192.168.1.12 dtmf-relay rtp-nte no vad ! dial-peer voice 1800 pots service session destination-pattern 18[0,6-8]+...$ no digit-strip port 2/0:23 ! sip-ua no remote-party-id retry invite 3 retry response 3 retry bye 3 retry cancel 3 timers trying 1000 timers buffer-invite 5000 sip-server ipv4:192.168.1.11 This router is currently running IOS 12.4, but the config was the same for 12.3T. I hope this helps. B. J. From: Jerry James [mailto:[EMAIL PROTECTED] Sent: Thursday, October 13, 2005 14:26 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Sample cisco config for cisco 7206 I see a lot of comments but no actual show runs. Can someone post a 7206 config. I am having a dickens of a time getting calls to pass. I currently have the following loaded. Cisco IOS Software, 7200 Software (C7200-IK9O3S-M), Version 12.3(8)T6, RELEASE SOFTWARE (fc2) Thanks !!! Jerry -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.344 / Virus Database: 267.11.14/130 - Release Date: 10/12/2005 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk and Dialogic
Hi all, I have a colleague who is very stuck on dialogic boards. I now the asterisk web site says it supports some dialogic boards but has anyone actually installed one and gotten it to work. I tried once to install Dialogic SR 5.1.1 with a D/41JCT-LS but gave up and ended up reformatting and going to a wildcard. I appreciate any feedback, as it will end up being my job to install and configure the server and I am not looking forward to it. Shawn ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Hang up problem Costa Rica Indications
Hi I have a asterisk working in Costa Rica and everything is working well except when an incoming call from the PSTN hangs up, asterisk wont hang up. The port is busy I probe the brazil configuration, but not work. Any ideas? , Olger Merlos V. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Middle Ground between POTS and T1?
I use a partial T1 as well (12B + 1D). Most CLECs offer them. -- Tom On 10/18/05, Goran Skular [EMAIL PROTECTED] wrote: Here (Croatia) is also possible to get partial E1 (PRI/PRA)... from one of our telcos (DT T-com) we can get PRA in 10 increments: 10B, 20B and 30B We have a partial T1 (5B + D, iirc) from Allstream - there may be a provider in your area that does something similar. Regards, -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web: http://www.dnv.org RSS Feed: http://www.dnv.org/rss.asp On 17-Oct-05, at 12:48 PM, Matthew T. O'Connor wrote: I was wondering if there was a middle ground between POTS lines and a T1. I have a new office with a T1 line and while it's working well, it's a lot of money and we will never use anywhere near 23 lines at one time. Is it possible to get a few ISDN lines or something and bundle them together? Basically I would like to get the digital features of the T1 PRI (DID number, etc...) but smaller. Thanks, Matthew ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tom ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Queues and call waiting indication
Hello, you should use asterisk agents and you'll see that the problem will go away. Bye l. On Tue, 18 Oct 2005 14:13:32 +0200, [EMAIL PROTECTED] wrote: Hi, I'm running 1.2 beta1 in a mini call center. I have 3 queues with 10 operators, and I'm running into some trouble because when all the operators are busy answering call asterisk still sends them more, resulting in a beep beep (call waiting) over and over again in Xlite audio. An easy solution woud be the use of a single line user agent, like firefly, still this behaviour does not make any sense to me. I tried using incominglimit and outgoinglimit in my sip.conf, even if they are deprecated: no luck. Here is a sample from my queues.conf, something wrong in my setup maybe ? Tnx for any help! [ingombranti] joinempty = strict maxlen=3 musiconhold = default announce = annuncio-ingombranti strategy = rrmemory servicelevel = 60 timeout = 15 announce-frequency = 15 eventwhencalled = yes member=SIP/401 member=SIP/402 member=SIP/403 member=SIP/404 member=SIP/405 member=SIP/406 member=SIP/407 member=SIP/408 member=SIP/409 member=SIP/410 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Loway Research - Home of QueueMetrics http://queuemetrics.loway.it ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] setting a dialplan on a GXP-2000 Grandstream
Not that I've seen.. about all you can do is adjust the inter digit timeout.. Louis-David Mitterrand wrote: Hi, I looked at the docs and probably missed it: is there a way to set a dialplan on the GXP-2000? (to avoid having to press Send) Thanks, begin:vcard fn:Matt Hess n:Hess;Matt org:LiveWire Networks adr;dom:;;4577 Pecos St;Denver;CO;80211 email;internet:[EMAIL PROTECTED] title:Sr. Network Engineer tel;work:303-458-5667 x 106 tel;fax:303-458-5725 x-mozilla-html:FALSE url:http://www.livewirenet.com/ version:2.1 end:vcard ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fax2Mail
Hello, Is there or can anyone provide a comprehensive guide (designed for Linux/Asterisk novices) to installing/setting upAsterisk in order to support Fax2Mail service? In my case, I would like Asterisk to receive fax calls to predefined numbers (ranges) and to associate each of these numbers to email addresses. Thank you in advance. David Yahoo! Music Unlimited - Access over 1 million songs. Try it free.___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Can IAX be used without going thre a IAX server
Is it possible to route a call from an Asterisk box through the Internet to a IAX device (in this case Digium IAXy) without using an IAX service like IAXTel? I have it working on my local Ethernet LAN so it should be possible to use VPN to cross the internet. Anyone using VPN or other method to acomplish this? Thanks Chad ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Recomendations for utility to generateAsteriskconfiguration
On Oct 19, 2005, at 9:26 AM, asterisk wrote:AMP's dialplan and setup is quite complex. Requires, e.g, a number ofAGIs.This is normally not the type of thing you'd like to hand-edit laterafter the initial adaptation to the target system.Who said anything about hand editing?That is why you would want to keep the old computer running [EMAIL PROTECTED] Insteadof hand editing anything, make the changes on the [EMAIL PROTECTED] box's AMP GUI andcopy them over again. Very simple and most tech folks have an old computerlaying around somewhere that could be put to use.Why wouldn't you just install [EMAIL PROTECTED] on your main server then? Why install a second server and go through the trouble of using scp to copy files back and forth?Tom___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Error on AAHome Beta 4
Hi, I have installed AAH beta 4 and I am getting this error. I have installed it from aahbeta.tar.gz so I can make the server dual boot. this is what I am getting in error, any clue how I can fix this? Thanks Warning: fopen(/etc/asterisk/vm_general.inc): failed to open stream: Permission denied in /var/www/html/admin/functions.php on line 2292 Warning: fopen(/etc/asterisk/vm_email.inc): failed to open stream: Permission denied in /var/www/html/admin/functions.php on line 2292 Warning: fopen(/etc/asterisk/voicemail.conf): failed to open stream: Permission denied in /var/www/html/admin/functions.php on line 2292 Warning: fopen(/etc/asterisk/vm_general.inc): failed to open stream: Permission denied in /var/www/html/admin/functions.php on line 2292 Warning: fopen(/etc/asterisk/vm_email.inc): failed to open stream: Permission denied in /var/www/html/admin/functions.php on line 2292 Warning: fopen(/etc/asterisk/voicemail.conf): failed to open stream: Permission denied in /var/www/html/admin/functions.php on line 2292 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can IAX be used without going thre a IAX server
Chadwick E. Labno wrote: Is it possible to route a call from an Asterisk box through the Internet to a IAX device (in this case Digium IAXy) without using an IAX service like IAXTel? I have it working on my local Ethernet LAN so it should be possible to use VPN to cross the internet. Anyone using VPN or other method to acomplish this? Yes. I do IAX2 to IAX2 all the time using either IPSec or GRE (Usually both). David ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Assistance with loging a particular event.
I am attempting to unify how numbers come to me from a specific T1, this T1 acts as an ingress for about 4000 DIDS. About 98% of those DIDS come in as a 10-digit DNIS, what I would like to do is have asterisk log when a number comes in 7 or 11 digit so I can contact my upstream provider and have them translate those DIDS to 10 digit. I think I can achieve this by modifying my dial-plan I am just hoping somebody else has done something similar so I dont have to re-create the wheel. Thanks in advance! Chris ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Recomendations for utility togenerateAsteriskconfiguration
AMP's dialplan and setup is quite complex. Requires, e.g, a number of AGIs. This is normally not the type of thing you'd like to hand-edit later after the initial adaptation to the target system. Who said anything about hand editing? That is why you would want to keep the old computer running [EMAIL PROTECTED]. Instead of hand editing anything, make the changes on the [EMAIL PROTECTED] box's AMP GUI and copy them over again. Very simple and most tech folks have an old computer laying around somewhere that could be put to use. Why wouldn't you just install [EMAIL PROTECTED] on your main server then? Why install a second server and go through the trouble of using scp to copy files back and forth? Tom Maybe because you snipped the beginning of the thread without reading the entire thread's context, but he is running on Solaris. I am not sure what all is involved with installing [EMAIL PROTECTED] on solaris but I assume it is no trivial task. WinSCP is very trivial IMHO and there is no "copying files back and forth", just one direction, takes about twenty seconds and maybe 30 if you are slow. Now you also have an almost hot swap server in case the Solaris machine goes down, just swap IP addresses and hardware. Dont think of it as a second server since it will have no clients (a server must have clients or it's not a server, right?) just think of it as a configuration generator. A good analogy is MS Frontpage. It is very common to use a graphical webdesign program to generate files of code (HTML) and then upload those files to your server. Same thing here. Thanks, Steve Totaro ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Error on AAHome Beta 4
Hi, I have installed AAH beta 4 and I am getting this error. I have installed it from aahbeta.tar.gz so I can make the server dual boot. this is what I am getting in error, any clue how I can fix this? Thanks Warning: fopen(/etc/asterisk/vm_general.inc): failed to open stream: Permission denied in /var/www/html/admin/functions.php on line 2292 Warning: fopen(/etc/asterisk/vm_email.inc): failed to open stream: Permission denied in /var/www/html/admin/functions.php on line 2292 Warning: fopen(/etc/asterisk/voicemail.conf): failed to open stream: Permission denied in /var/www/html/admin/functions.php on line 2292 Warning: fopen(/etc/asterisk/vm_general.inc): failed to open stream: Permission denied in /var/www/html/admin/functions.php on line 2292 Warning: fopen(/etc/asterisk/vm_email.inc): failed to open stream: Permission denied in /var/www/html/admin/functions.php on line 2292 Warning: fopen(/etc/asterisk/voicemail.conf): failed to open stream: Permission denied in /var/www/html/admin/functions.php on line 2292 ___ Try chowning those files to asterisk. I also think there is a script that changes file ownership in the /var/aah_build directory (i am guessing here) Thanks, Steve Totaro ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] select codec based on extension
I've the following installation :
|asterisk client| --- |asterisk server| --- |other asterisk server|
all the connections are made in IAX, the client and first server allows
711 and 729
the other server only allows 729 since it has low bandwidth at disposal
all the numbers but a few are routed to a digium card in the first
server, the others are routed to the other server, this way :
[default]
exten = _123X.,1,Dial(IAX2/otherserver/${EXTEN})
exten = _123X.,2,Hangup
exten = _X.,1,Dial(Zap/g1/${EXTEN})
exten = _X.,2,Hangup
when I call 123456 from the client box ...
on the client :
Call accepted by asterisk server (format alaw)
on the server :
Call accepted by other asterisk server (format g729)
on the other server :
Called [EMAIL PROTECTED]
and then on the server in the middle :
Oct 18 18:00:37 NOTICE[2846]: channel.c:1724 ast_set_write_format:
Unable to find a path from alaw to g729
Oct 18 18:00:37 NOTICE[2846]: channel.c:1757 ast_set_read_format: Unable
to find a path from g729 to alaw
since that something at the end of the call and the paps which sits
before the first asterisk server both have g729, I don't like too much
having to pay to translate something which need not translation.
Is there a clever combination of sip.conf, iax.conf and extensions.conf
I'm missing to solve my problem ?
___
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Re: [Asterisk-Users] Uniden UIP200 Issues
Jeff Herring wrote: Phone won't register on LAN port registers but doesn't work on PC port. SIP to SIP works. Anyone have a Configuration that works out there? Phone has 4.63 Firmware Make sure you have nat=never (or nat=route). Regards, -- Jason Becker Director CEO Coalescent Systems Inc. Enabling Open Source Telephony 403.244.8089 www.coalescentsystems.ca ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can IAX be used without going thre a IAX server
Is it possible to route a call from an Asterisk box through the Internet to a IAX device (in this case Digium IAXy) without using an IAX service like IAXTel? I have it working on my local Ethernet LAN so it should be possible to use VPN to cross the internet. Anyone using VPN or other method to acomplish this? Thanks Chad ___ Yes. You could use VPN if you need it for other reasons or do not want to open ports on your firewall but if you dont mind opening ports and have no need for VPN there is a much easier solution. Go into your router and set port forwarding. Set port 4569 to forward to the IP of your * box. Then reconfigure your IAXy to point to the public IP of your router. This works great! I usually go a step further and setup a DNS record for my domain to point to the public IP such as iax2.yourdomain.com to make it easier to remember. Thanks, Steve Totaro ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Newbie Question: Help with incoming dial plan
Title: Newbie Question: Help with incoming dial plan Hi all. I just got Asterisk installed with a Digium TE110P T1 card. Have it working for outbound calls so I know that all the hardware is functioning. Since all inbound calls come through my T1, I would like to setup a dial plan that handles the incoming call and tells the caller to enter the extension they wish to reach. All of my real extensions are in the [default] context, and the Zaptel is configured to go to the [default-incoming] context. It is the [default-incoming] context that I am unsure of how to set. If anyone could provide some insight, it would be much appreciated.get me started at least. David A. Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodata.net Tel: (519) 951-6079 Fax: (519) 451-6615 Poor planning on your part does not necessarily constitute an emergency on my part! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Newbie Question: Help with incoming dial plan
Title: Newbie Question: Help with incoming dial plan This is how I do it. [default-incoming]exten = 2691,1,Goto(extensions,3212,1)exten = 2692,1,Goto(extensions,3204,1)exten = 2693,1,Goto(extensions,3207,1)exten = 2694,1,Goto(extensions,3212,1)exten = 2695,1,Goto(extensions,3205,1)exten = 2696,1,Goto(extensions,3208,1)exten = 2697,1,Goto(extensions,1105,1)exten = 3211,1,Goto(extensions,,1)exten = 3223,1,Goto(extensions,3207,1) You will have to know how many digits are being sent, in my case it is four. So for example, someone dials the DID xxx-xxx-2691 the dialplan matches on the first line and then sends the cal to the context "extensions" (you would replace with default) and the extension "3212" in the first priority. If more or less digits are being sent by the telco, you will have to adjust the exten = to match. Sometimes they send three. Thanks, Steve Totaro - Original Message - From: Dave Morrow To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Tuesday, October 18, 2005 11:26 AM Subject: [Asterisk-Users] Newbie Question: Help with incoming dial plan Hi all. I just got Asterisk installed with a Digium TE110P T1 card. Have it working for outbound calls so I know that all the hardware is functioning. Since all inbound calls come through my T1, I would like to setup a dial plan that handles the incoming call and tells the caller to enter the extension they wish to reach. All of my real extensions are in the [default] context, and the Zaptel is configured to go to the [default-incoming] context. It is the [default-incoming] context that I am unsure of how to set. If anyone could provide some insight, it would be much appreciated .get me started at least. David A. Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodata.net Tel: (519) 951-6079 Fax: (519) 451-6615 Poor planning on your part does not necessarily constitute an emergency on my part! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message.Checked by AVG Anti-Virus.Version: 7.0.344 / Virus Database: 267.12.1/136 - Release Date: 10/15/2005 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Newbie Question: Help with incoming dial plan
Title: Newbie Question: Help with incoming dial plan I do not use any DID, all calls come in on the same number 111222 so what I would like to do is simply prompt the caller to enter the extension they wish to reach, then redirect to that extension in the [default] context. David A. Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodata.net Tel: (519) 951-6079 Fax: (519) 451-6615 Poor planning on your part does not necessarily constitute an emergency on my part! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of asteriskSent: Wednesday, October 19, 2005 11:34 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Newbie Question: Help with incoming dial plan This is how I do it. [default-incoming]exten = 2691,1,Goto(extensions,3212,1)exten = 2692,1,Goto(extensions,3204,1)exten = 2693,1,Goto(extensions,3207,1)exten = 2694,1,Goto(extensions,3212,1)exten = 2695,1,Goto(extensions,3205,1)exten = 2696,1,Goto(extensions,3208,1)exten = 2697,1,Goto(extensions,1105,1)exten = 3211,1,Goto(extensions,,1)exten = 3223,1,Goto(extensions,3207,1) You will have to know how many digits are being sent, in my case it is four. So for example, someone dials the DID xxx-xxx-2691 the dialplan matches on the first line and then sends the cal to the context "extensions" (you would replace with default) and the extension "3212" in the first priority. If more or less digits are being sent by the telco, you will have to adjust the exten = to match. Sometimes they send three. Thanks, Steve Totaro - Original Message - From: Dave Morrow To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Tuesday, October 18, 2005 11:26 AM Subject: [Asterisk-Users] Newbie Question: Help with incoming dial plan Hi all. I just got Asterisk installed with a Digium TE110P T1 card. Have it working for outbound calls so I know that all the hardware is functioning. Since all inbound calls come through my T1, I would like to setup a dial plan that handles the incoming call and tells the caller to enter the extension they wish to reach. All of my real extensions are in the [default] context, and the Zaptel is configured to go to the [default-incoming] context. It is the [default-incoming] context that I am unsure of how to set. If anyone could provide some insight, it would be much appreciated.get me started at least. David A. Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodata.net Tel: (519) 951-6079 Fax: (519) 451-6615 Poor planning on your part does not necessarily constitute an emergency on my part! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message.Checked by AVG Anti-Virus.Version: 7.0.344 / Virus Database: 267.12.1/136 - Release Date: 10/15/2005 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Newbie Question: Help with incoming dial plan
Title: Newbie Question: Help with incoming dial plan exten = s,1,Answerexten = s,2,Wait,2exten = s,3,Background(enter-ext-of-person)exten = s,4,DigitTimeout,5exten = s,5,ResponseTimeout,10 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dave MorrowSent: 18 October 2005 16:41To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [Asterisk-Users] Newbie Question: Help with incoming dial plan I do not use any DID, all calls come in on the same number 111222 so what I would like to do is simply prompt the caller to enter the extension they wish to reach, then redirect to that extension in the [default] context. David A. Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodata.net Tel: (519) 951-6079 Fax: (519) 451-6615 Poor planning on your part does not necessarily constitute an emergency on my part! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of asteriskSent: Wednesday, October 19, 2005 11:34 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Newbie Question: Help with incoming dial plan This is how I do it. [default-incoming]exten = 2691,1,Goto(extensions,3212,1)exten = 2692,1,Goto(extensions,3204,1)exten = 2693,1,Goto(extensions,3207,1)exten = 2694,1,Goto(extensions,3212,1)exten = 2695,1,Goto(extensions,3205,1)exten = 2696,1,Goto(extensions,3208,1)exten = 2697,1,Goto(extensions,1105,1)exten = 3211,1,Goto(extensions,,1)exten = 3223,1,Goto(extensions,3207,1) You will have to know how many digits are being sent, in my case it is four. So for example, someone dials the DID xxx-xxx-2691 the dialplan matches on the first line and then sends the cal to the context "extensions" (you would replace with default) and the extension "3212" in the first priority. If more or less digits are being sent by the telco, you will have to adjust the exten = to match. Sometimes they send three. Thanks, Steve Totaro - Original Message - From: Dave Morrow To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Tuesday, October 18, 2005 11:26 AM Subject: [Asterisk-Users] Newbie Question: Help with incoming dial plan Hi all. I just got Asterisk installed with a Digium TE110P T1 card. Have it working for outbound calls so I know that all the hardware is functioning. Since all inbound calls come through my T1, I would like to setup a dial plan that handles the incoming call and tells the caller to enter the extension they wish to reach. All of my real extensions are in the [default] context, and the Zaptel is configured to go to the [default-incoming] context. It is the [default-incoming] context that I am unsure of how to set. If anyone could provide some insight, it would be much appreciated.get me started at least. David A. Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodata.net Tel: (519) 951-6079 Fax: (519) 451-6615 Poor planning on your part does not necessarily constitute an emergency on my part! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message.Checked by AVG Anti-Virus.Version: 7.0.344 / Virus Database: 267.12.1/136 - Release Date: 10/15/2005 NOTICE: This e-mail message and all attachments transmitted with it may contain legally privileged and confidential information intended solely for the use of the addressee. If the
Re: [Asterisk-Users] Newbie Question: Help with incoming dial plan
Title: Newbie Question: Help with incoming dial plan add this context [default-incoming]exten = 111222,1,Goto(default-incoming,s,1) exten = s,1,Answerexten = s,2,DigitTimeout(10)exten = s,3,ResponseTimeout(20)exten = s,4,Background(swelcome)exten = t,1,Hangupinclude = extensions add this to your extensions context ;directory appexten = 9,1,Directory(default-extensions) ; exten for recording greetings/menusexten = 12,1,Authenticate(1234|)exten = 12,2,Wait(2)exten = 12,3,Record(/var/lib/asterisk/sounds/welcome:gsm)exten = 12,4,Wait(2)exten = 12,5,Playback(/var/lib/asterisk/sounds/welcome)exten = 12,6,Wait(2)exten = 12,7,Hangup Reload and dial 12 with the password of 1234 and record your greeting and then hangup. If you mess up just do it over. Thanks, Steve - Original Message - From: Dave Morrow To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Tuesday, October 18, 2005 11:41 AM Subject: RE: [Asterisk-Users] Newbie Question: Help with incoming dial plan I do not use any DID, all calls come in on the same number 111222 so what I would like to do is simply prompt the caller to enter the extension they wish to reach, then redirect to that extension in the [default] context. David A. Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodata.net Tel: (519) 951-6079 Fax: (519) 451-6615 Poor planning on your part does not necessarily constitute an emergency on my part! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of asteriskSent: Wednesday, October 19, 2005 11:34 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Newbie Question: Help with incoming dial plan This is how I do it. [default-incoming]exten = 2691,1,Goto(extensions,3212,1)exten = 2692,1,Goto(extensions,3204,1)exten = 2693,1,Goto(extensions,3207,1)exten = 2694,1,Goto(extensions,3212,1)exten = 2695,1,Goto(extensions,3205,1)exten = 2696,1,Goto(extensions,3208,1)exten = 2697,1,Goto(extensions,1105,1)exten = 3211,1,Goto(extensions,,1)exten = 3223,1,Goto(extensions,3207,1) You will have to know how many digits are being sent, in my case it is four. So for example, someone dials the DID xxx-xxx-2691 the dialplan matches on the first line and then sends the cal to the context "extensions" (you would replace with default) and the extension "3212" in the first priority. If more or less digits are being sent by the telco, you will have to adjust the exten = to match. Sometimes they send three. Thanks, Steve Totaro - Original Message - From: Dave Morrow To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Tuesday, October 18, 2005 11:26 AM Subject: [Asterisk-Users] Newbie Question: Help with incoming dial plan Hi all. I just got Asterisk installed with a Digium TE110P T1 card. Have it working for outbound calls so I know that all the hardware is functioning. Since all inbound calls come through my T1, I would like to setup a dial plan that handles the incoming call and tells the caller to enter the extension they wish to reach. All of my real extensions are in the [default] context, and the Zaptel is configured to go to the [default-incoming] context. It is the [default-incoming] context that I am unsure of how to set. If anyone could provide some insight, it would be much appreciated .get me started at least. David A. Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodata.net Tel: (519) 951-6079 Fax: (519) 451-6615 Poor planning on your part does not necessarily constitute an emergency on my part! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ___--Bandwidth and Colocation sponsored by
Re: [Asterisk-Users] compiling Asterisk 1.2 with zaptel and h.323
On FC4 is better to use pwlib 1.9.1 and openh323 1.17.2. I think, that OPENH3232DIR= is wrong. Better is OPENH323DIR= :-). If You use standard prefix for instalation o packages there is a better way instad copy library edit /etc/ld.co.conf and use /usr/local/lib/ as next source of shared library. Anyway, your text is very usefull. Bob. Dne pondělí 17 říjen 2005 14:55 Lenz napsal(a): Hello list, I have prepared a small recipe on how to compile Asterisk 1.2 beta 1 with a TDM400 card and H.323. You can find it at http://www.oinko.net/astrecipes/index.php?n=102 Any comment / suggestion / modification /bugfix is welcome! I was wondering: is there any way to build a version of Bristuff for 1.2 beta 1? Bye for now, l. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Polycom IP501 and record on demand
I am doing some experimenting with Asterisk 1.0.9 and Polycom IP501's. I have the extensions setup, and everything is working well up to this point. Now, I want to setup my system so that a user at an extension can start a recording on demand. I have tried various Google searches, but am coming up empty. Could someone point me in the right direction, or have some sample Polycom/Asterisk configuration files I can look at? Ideally, I would like this to just be programmed to a soft key on the IP501. Thanks, James ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Error on AAHome Beta 4
They are chown to asterisk:asterisk and chmod 777 . But I am still getting those error. Any other suggestion? Thanks Quoting asterisk [EMAIL PROTECTED]: Hi, I have installed AAH beta 4 and I am getting this error. I have installed it from aahbeta.tar.gz so I can make the server dual boot. this is what I am getting in error, any clue how I can fix this? Thanks Warning: fopen(/etc/asterisk/vm_general.inc): failed to open stream: Permission denied in /var/www/html/admin/functions.php on line 2292 Warning: fopen(/etc/asterisk/vm_email.inc): failed to open stream: Permission denied in /var/www/html/admin/functions.php on line 2292 Warning: fopen(/etc/asterisk/voicemail.conf): failed to open stream: Permission denied in /var/www/html/admin/functions.php on line 2292 Warning: fopen(/etc/asterisk/vm_general.inc): failed to open stream: Permission denied in /var/www/html/admin/functions.php on line 2292 Warning: fopen(/etc/asterisk/vm_email.inc): failed to open stream: Permission denied in /var/www/html/admin/functions.php on line 2292 Warning: fopen(/etc/asterisk/voicemail.conf): failed to open stream: Permission denied in /var/www/html/admin/functions.php on line 2292 ___ Try chowning those files to asterisk. I also think there is a script that changes file ownership in the /var/aah_build directory (i am guessing here) Thanks, Steve Totaro ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users CM Rahman Jr. CTO CCS Internet www.ccsi.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hints on hardware to use
On Wednesday 19 October 2005 15:34, asterisk wrote: Hello, I have to deploy an Asterisk PBX with this requirements: - 1 or 2 ISDN lines in input/output - 14 internal analog phones (yes, I know, analog ones... ;( ) - Billing interface for the operator (for usage of analog phones) For the external interface I'm thinking about Beronet Quad Span ISDN/BRI Card TE/NT (BN4S0). I'm in trouble about the internal interfaces: the first thought was about Digium card (4 x TDM400P with 2 x S100M modules each), but it's difficult to find a MB with 5 PCI, and I'll have no chanches for future expansions. Does anyone of you know a PCI card with 8 FXS port that SURELY works with Asterisk? I'm ready to examine any other piece of hardware with 8 or more FXS ports, too... By the way, for billing operations I'm going to check AstBill sofware; did anyone positively try it with asterisk in operational environment? Jonathan For your internal analog extensions why not get a Digium T1 card and a channel bank. I only have experience with Adtran 600E but they work extremely well and can be had used on ebay for about $600 regularly (if you are lucky you may be able to get it much cheaper.) I read the Rhino channel banks are much cheaper and work well with asterisk but have no personal experience. With the Channel bank solution, you are looking at $500 for the T1 board and another $600 for the channel bank with 24 FXS ports. Its a solid solution and gives you tons of room to upgrade from 14 FXS ports to 24 by simply adding phones. Thanks, Steve Totaro It's surely a better way than mine to solve the problem... But how can I integrate Adtran 600E with Asterisk box (apart the phisical connection with the two T1 ports)? How changes the configuration of Asterisk files? Is it like a bridge across two T1 lines or what? I'm not so expert about this type of Asterisk configurations: can I find hints or docs somewhere? I've also googled a bit, and I've found the MOSA3716 box: 16 FXS and 2 ethernet ports, for about 1.200$ at bobascom. What do you think about it? It seems * compatible, and with ethernet ports it wouldn't need anything else than 1 ISDN card for inbound/outgoing calls... It could be completely transparent to * box and the analog phones... Have you ever heard something about it? Thanks in advance Jonathan ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: PRI echo issues: solvable?
Andrew Kohlsmith [EMAIL PROTECTED] wrote: I've never seen that, it's always when we call out. Certain numbers will always trigger it. 888-737-4787 (IPC Resistors, it dumps into an IVR so it's safe to call) is one such number, but we have local numbers that hit other I just tried this number, and it was answered by a person. Doug -- Doug Meredith ([EMAIL PROTECTED]) SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Redundency
Hi, I wish to use Asterisk as a SIP server. How do I use Asterisk in a redundent network? So, if one Asterisk server fails, how does failover work? James ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Vontage Problems
Has anyone experienced problems with Vontage and Asterisk. I'm using Asterisk (Current Stable) and Sipura-841 phones.While talking on an outbound call the transmission seems to fade out until the other person can't hear me but I can hear them. I've updated the firmware on the 841 but it had no effect. I've also tested the phones on another server using Teliax for termination and I have not had any trouble. Regards, Chris___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fwd: {100-1287} RE: DIDs
Note: forwarded message attached.---BeginMessage--- Someone will be in contact with you within the next couple of hours to discuss your account. Regards, I am Jerry Richmond CEO of ByVolution LLC. We have purched some did's from you that we use to test with, weare going to order our first batch of 250 this week. John Blackman is not with us any more. I need for some one to call me on my cell phone because our office no. 9198270713 to 9198270720 can not except outside calls. Brian Sponaugle tells me he is not getting the help he needs from Sellvoip I know something is wrong. We will over 10,000 did in the next 12 months. Help me do bussiness with you. Jerry F. Richmond 8606 Jersey Court Raleighj, NC 27617 919 606 7685 cell 919 827 0714 work phone (not Working) Try it. ---End Message--- ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fwd: {100-1287} RE: DIDs
I will be on my cell 919 606 7685.
We need help bad.Jerry Richmond [EMAIL PROTECTED] wrote:
Note: forwarded message attached.Date: Tue, 18 Oct 2005 11:04:09 GMTTo: [EMAIL PROTECTED]From: "Sales Support" [EMAIL PROTECTED]Subject: {100-1287} RE: DID"sSomeone will be in contact with you within the next couple of hours to discuss your account.Regards, I am Jerry Richmond CEO of ByVolution LLC. We have purched some did's from you that we use to test with, weare going to order our first batch of 250 this week. John Blackman is not with us any more. I need for some one to call me on my cell phone because our office no. 9198270713 to 9198270720 can not except outside calls. Brian Sponaugle tells me he is not getting the help he needs from Sellvoip I know something is wrong. We will over 10,000 did in the next 12 months. Help me do bussiness with you. Jerry F. Richmond 8606 Jersey Court Raleighj, NC 27617 919 606 768
5
cell 919 827 0714 work phone (not Working) Try it.___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users___
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[Asterisk-Users] sip rfc bye violated?
I have this in sip show history for a particular channel marked as dead (should be removed) in sip show channels: 1. TxReqRelINVITE / 102 INVITE 2. Rx SIP/2.0 / 102 INVITE 3. CancelDestroy 4. Rx SIP/2.0 / 102 INVITE 5. CancelDestroy 6. Unhold SIP/2.0 7. Rx SIP/2.0 / 102 INVITE 8. CancelDestroy 9. Unhold SIP/2.0 10. Rx SIP/2.0 / 102 INVITE 11. CancelDestroy 12. Unhold SIP/2.0 13. TxReq ACK / 102 ACK 14. TxReqRelINVITE / 103 INVITE 15. Rx SIP/2.0 / 103 INVITE 16. CancelDestroy 17. Rx SIP/2.0 / 103 INVITE 18. CancelDestroy 19. Unhold SIP/2.0 20. TxReq ACK / 103 ACK 21. TxReqRelINVITE / 104 INVITE 22. Rx BYE / 302 BYE 23. TxResp SIP/2.0 / 302 BYE 24. Rx SIP/2.0 / 104 INVITE 25. CancelDestroy Why is asterisk allowing an invite after receiving a bye on a particular session/channel? From what I've read.. a bye should be the termination of the session/channel and therefore it should be hungup and removed.. yet it is not. I am using cvs head from 2005-10-08 00:00 .. I can't use the latest cvs head as it's rather ugly with sip right now.. especially on refer/redirect/reinvites.. but that will be left for a different topic. I believe from looking at things that the sip gateway involved with the sip session is re-using a particular call identifier immediately after it believes that call from before is gone.. (possibly a bug on the vendor side as far as that goes) but regardless of whether the vendor is immediately re-using a session id or not should not matter as the fact seems to be that asterisk allows this situation to happen when (from what I've been reading) it should not. Does anyone have any comments or thoughts on this? begin:vcard fn:Matt Hess n:Hess;Matt org:LiveWire Networks adr;dom:;;4577 Pecos St;Denver;CO;80211 email;internet:[EMAIL PROTECTED] title:Sr. Network Engineer tel;work:303-458-5667 x 106 tel;fax:303-458-5725 x-mozilla-html:FALSE url:http://www.livewirenet.com/ version:2.1 end:vcard ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fwd: {100-1287} RE: DIDs
Jerry if you have something to ask or say about your vendor on this list do so. But please stop dumping a copy here of all communications with them. Jerry Richmond wrote: Note: forwarded message attached. big snip ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Newbie Question: Help with incoming dial plan
Title: Newbie Question: Help with incoming dial plan Thanks Steve, this works like a charm! Might I ask how I setup that Directory? David A. Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodata.net Tel: (519) 951-6079 Fax: (519) 451-6615 Poor planning on your part does not necessarily constitute an emergency on my part! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of asteriskSent: Wednesday, October 19, 2005 11:51 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Newbie Question: Help with incoming dial plan add this context [default-incoming]exten = 111222,1,Goto(default-incoming,s,1) exten = s,1,Answerexten = s,2,DigitTimeout(10)exten = s,3,ResponseTimeout(20)exten = s,4,Background(swelcome)exten = t,1,Hangupinclude = extensions add this to your extensions context ;directory appexten = 9,1,Directory(default-extensions) ; exten for recording greetings/menusexten = 12,1,Authenticate(1234|)exten = 12,2,Wait(2)exten = 12,3,Record(/var/lib/asterisk/sounds/welcome:gsm)exten = 12,4,Wait(2)exten = 12,5,Playback(/var/lib/asterisk/sounds/welcome)exten = 12,6,Wait(2)exten = 12,7,Hangup Reload and dial 12 with the password of 1234 and record your greeting and then hangup. If you mess up just do it over. Thanks, Steve - Original Message - From: Dave Morrow To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Tuesday, October 18, 2005 11:41 AM Subject: RE: [Asterisk-Users] Newbie Question: Help with incoming dial plan I do not use any DID, all calls come in on the same number 111222 so what I would like to do is simply prompt the caller to enter the extension they wish to reach, then redirect to that extension in the [default] context. David A. Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodata.net Tel: (519) 951-6079 Fax: (519) 451-6615 Poor planning on your part does not necessarily constitute an emergency on my part! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of asteriskSent: Wednesday, October 19, 2005 11:34 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Newbie Question: Help with incoming dial plan This is how I do it. [default-incoming]exten = 2691,1,Goto(extensions,3212,1)exten = 2692,1,Goto(extensions,3204,1)exten = 2693,1,Goto(extensions,3207,1)exten = 2694,1,Goto(extensions,3212,1)exten = 2695,1,Goto(extensions,3205,1)exten = 2696,1,Goto(extensions,3208,1)exten = 2697,1,Goto(extensions,1105,1)exten = 3211,1,Goto(extensions,,1)exten = 3223,1,Goto(extensions,3207,1) You will have to know how many digits are being sent, in my case it is four. So for example, someone dials the DID xxx-xxx-2691 the dialplan matches on the first line and then sends the cal to the context "extensions" (you would replace with default) and the extension "3212" in the first priority. If more or less digits are being sent by the telco, you will have to adjust the exten = to match. Sometimes they send three. Thanks, Steve Totaro - Original Message - From: Dave Morrow To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Tuesday, October 18, 2005 11:26 AM Subject: [Asterisk-Users] Newbie Question: Help with incoming dial plan Hi all. I just got Asterisk installed with a Digium TE110P T1 card. Have it working for outbound calls so I know that all the hardware is functioning. Since all inbound calls come through my T1, I would like to setup a dial plan that handles the incoming call and tells the caller to enter the extension they wish to reach. All of my real extensions are in the [default] context, and the Zaptel is configured to go to the [default-incoming]
[Asterisk-Users] Re: Vontage Problems
I am a newbie and want to step up to VoIP and switch from analog connetion to my Astrisk/Lineox box. Any suggestions on configuring Vontage and what to get/ask when signing up? Has anyone experienced problems with Vontage and Asterisk. I'm using Asterisk (Current Stable) and Sipura-841 phones.While talking on an outbound call the transmission seems to fade out until the other person can't hear me but I can hear them. I've updated the firmware on the 841 but it had no effect. I've also tested the phones on another server using Teliax for termination and I have not had any trouble. Regards, Chris ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem loading misdn driver
Hallo all, i have a problem on loading chan_misdn. The misdn is running and all cards (TDM40B+AVMFritz) is initialized. When im going to start asterisk with the chan_misdn.so module i get the following error in the log (on console) and asterisk ist hanging. i use the current CVS-HEAD of asterisk (7 Days old), chan_misdn-0.2.1-rc2 and mISDN+mISDNuser from the automated installation. Is here anybody who have any idea why asterisk hangs. I searched all the mailinglists, and doesn't get any information on what's wrong. cp from the console: [chan_local.so] = (Local Proxy Channel) == Registered channel type 'Local' (Local Proxy Channel Driver) [chan_skinny.so] = (Skinny Client Control Protocol (Skinny)) == Parsing '/etc/asterisk/skinny.conf': Found == Skinny listening on 0.0.0.0:2000 == Registered channel type 'Skinny' (Skinny Client Control Protocol (Skinny)) [chan_features.so] = (Feature Proxy Channel) == Registered channel type 'Feature' (Feature Proxy Channel Driver) [skipping chan_oss.so] [chan_modem_i4l.so] = (ISDN4Linux Emulated Modem Driver) [chan_phone.so] = (Linux Telephony API Support) == Parsing '/etc/asterisk/phone.conf': Found == Registered channel type 'Phone' (Standard Linux Telephony API Driver) [chan_oh323.so] = (InAccess Networks OpenH323 Channel Driver) == Parsing '/etc/asterisk/rtp.conf': Found == Parsing '/etc/asterisk/oh323.conf': Found == Registered channel type 'OH323' (InAccess Networks OpenH323 Channel Driver) == OpenH323 Channel Ready (v0.7.3) [chan_misdn.so] = (Channel driver for mISDN Support (Bri/Pri)) cb_log called with out-of-range port number! Yours Hans-Peter Straub -- ---* I-NetPartner GmbH Hans-Peter Straub Seewiesenstrasse 12 D-73054 Eislingen -- Phone: +49 7161 9849955 Fax: +49 7161 9849956 -- eMail: [EMAIL PROTECTED] Web: http://www.I-NetPartner.de ---* ** Informieren Sie Sich über ** -- GigaLan -- ** das Funknetz im Filstal ** http://www.GigaLan.de ---* -- PGP-ID: 24557EED PGP-Key: http://www.i-netpartner.de/hps.asc PGP-Fingerprint: 51F2 31E4 4361 1B7F 8648 60D9 FC1A 68D2 2455 7EED -- ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom IP501 and record on demand
Hi James, [EMAIL PROTECTED] wrote: I am doing some experimenting with Asterisk 1.0.9 and Polycom IP501's. I have the extensions setup, and everything is working well up to this point. Now, I want to setup my system so that a user at an extension can start a recording on demand. I have tried various Google searches, but am coming up empty. Could someone point me in the right direction, or have some sample Polycom/Asterisk configuration files I can look at? Ideally, I would like this to just be programmed to a soft key on the IP501. Thanks You could also take a look at features.conf, and use ** for blind transfers, ## for attended transfers, *0 for recording, and *1 to hangup. I haven't tried mapping them to polycom buttons, but there was recently a discussion about that, just this week you can search the archives. matt ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: PRI echo issues: solvable?
Subject: RE: [Asterisk-Users] PRI echo issues: solvable?
Kris Boutilier [EMAIL PROTECTED] wrote:
On Tuesday 11 October 2005 11:49, alan wrote:
After solving the other low hanging fruit audio issues in our Asterisk
PBX, we are left with occasional cases of severe echo which we have not
found a solution for yet.
{clip}
Most calls have minimal, acceptable echo levels. But occasionally, we
get a call where the echo is delayed by a substantial amount (sometimes
around 250ms), and sounds as loud as the remote party.
Yup.
I finally got time to spend on looking into this again.
If you were to use ztmonitor on the channel to record the transmit and
receive sides to separate wav files, drive an impulse down the channel
(ie. a sharp, loud click) and then load the files into a tool where
they could be viewed side by side you'd see the actual echo endpath
(tail) length.
How does one use ztmonitor to record into separate files for transmit
and receive? My ztmonitor man page doesn't describe how to do this, it
only allows one -f File specification.
When I monitored a sample conversation with ztmonitor, it recorded both
channels in one file. Then I set up a call from the number which gives
us big echo. Although I heard echo when I was on the call, the
recorded version of the call did not record the echo.
There are two possibilities here:
1. it wasn't recording the incoming call leg.
2. the echo is entirely internal to our system
I couldn't see how #1 could be the case when the immediately previous
ztmonitor recorded both call legs. On the other hand, I told the remote
party to just put the phone down so I have no evidence that any sound
they made was missing.
But I can't see how #2 is the case either, since it only affects certain
phone numbers, and it affects them consistently whenever it happens.
I have a feeling I just did something wrong, so I'll go back and try
again. I'll also try a greater echocancel= value and see if it
helps.
Thanks,
Alan Ferrency
pair Networks, Inc.
[EMAIL PROTECTED]
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Re: [Asterisk-Users] Asterisk Redundency
Hi, I wish to use Asterisk as a SIP server. How do I use Asterisk in a redundent network? So, if one Asterisk server fails, how does failover work? James James, I've been working on the same thing. I think it's pretty important too because phone providers shoot for five-nine availablity, so people are pretty much accustomed to their phones just plain working all the time. And if they don't work there's hell to pay. I think the simplest way would be to use SIP adapters that allow you to specify a backup server. I can't think of any off the top of my head, but I have definitely seen them. Then you just need to keep the configs synchronized between your two servers (via rsync or what have you). The problems of course are that it limits your options for SIP adapters, and you have to be in an evironment where you can control which SIP adapters people are using. Since I can't do that, what I've settled on is heartbeat + mon. Heartbeat will monitor for a system level failure and switch to the backup machine if neccesary; and mon will watch the asterisk (or any other) service and restart it and/or alert me if it fails. There are other options for high availability. Google for linux clustering or [your platform] clustering. I only stuck with heartbeat and mon because they were both relatively simple to set up (as long as you stick with a heartbeat version 1 config). ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dial command in extensions
Kevin Bockman wrote: Patrick wrote: is there anyway to make the dial command return and execute the next line in the dial plan after the channel hangs up? Try with h (for hangup): exten = 1234,1,Dial... exten = 1234,h,... He actually meant the 'h' exten and not priority: exten = h,1,blah yeah. i figured that. but that would execute on everything in the context. Someone else suggested the g option on Dial. that might work better. i'll have to experiment on it. thanks. -- Edwin Lam [EMAIL PROTECTED] Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Asterisk-Users Digest, Vol 15, Issue 108
Hi James - I am doing some experimenting with Asterisk 1.0.9 and Polycom IP501's. I have the extensions setup, and everything is working well up to this point. Now, I want to setup my system so that a user at an extension can start a recording on demand. I have tried various Google searches, but am coming up empty. Could someone point me in the right direction, or have some sample Polycom/ Asterisk configuration files I can look at? Ideally, I would like this to just be programmed to a soft key on the IP501. This sounds like another thread that was recently on this list. AFAIK, the only way to record a live conversation is to transfer a call to an extension with a monitor command. That requires at least two key presses (e.g. #8). The key remapping feature of the Polycom phones only allows you to map a single key to another key. However, some ingenious person on the list came up with the idea of mapping a speed dial to another key to handle multiple key presses mapped to a single key. Unfortunately, the Polycom phones always interpret a speed dial as request to start a new channel, so it can't be done on calls already in progress. There doesn't seem to be any kind of workaround yet. I'd love to be able to do something similar for call parking. We may have to petition Polycom for a new feature. If you do happen to come up with anything, though, let us know. - Noah ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] strange behavior after turning jitter buffer on
This is with asterisk 1.20beta1:
I was experiencing moments of sporadic silence, so I thought to turn on
the jitter buffer in iax.conf.
I started with the following settings, which are basically ripped from
the sample config:
jitterbuffer=yes
forcejitterbuffer=no
maxjitterbuffer=1000
resyncthreshold=1000
maxjitterinterps=10
After doing this I encountered something a little different. When I
call my cell phone from a SIP phone (via asterisk and an IAX
connection); the cell rings, I answer it, the cell claims it is
connected, but I continue to hear ringing on the SIP phone until the
Dial application times out (45 secs). I don't see anything bad
happening in the log except that chan_iax2 seems to think that no one
has answered. To make it more interesing, this doesn't happen on every
call.
Excerpt from the log file detailing the call is below (with my actual
cell phone number and teliax username obscured).
Does anyone have any thoughts on the matter?
Log stuff:
Oct 18 13:52:12 DEBUG[12842] chan_sip.c: * SIP extension value: 0 for
call e0dae
[EMAIL PROTECTED]
Oct 18 13:52:12 DEBUG[12842] chan_sip.c: Setting NAT on RTP to 0
Oct 18 13:52:12 DEBUG[12842] chan_sip.c: Stopping retransmission on
'e0dae938-16
[EMAIL PROTECTED]' of Response 101: Match Found
Oct 18 13:52:12 DEBUG[12842] chan_sip.c: * SIP extension value: 0 for
call e0dae
[EMAIL PROTECTED]
Oct 18 13:52:12 DEBUG[12842] chan_sip.c: Setting NAT on RTP to 0
Oct 18 13:52:12 DEBUG[12842] chan_sip.c: Checking SIP call limits for
device PLX
Fax
Oct 18 13:52:12 DEBUG[12842] chan_sip.c: build_route: Contact hop:
sip:[EMAIL PROTECTED]
68.215.99.200:5060
Oct 18 13:52:12 VERBOSE[12842] logger.c: -- Executing
SetCallerID(SIP/PLXFa
x-ceb5, 8667594678 |a) in new stack
Oct 18 13:52:12 VERBOSE[12842] logger.c: -- Executing
Dial(SIP/PLXFax-ceb5
, IAX2/[EMAIL PROTECTED]/1607999|45|Tr) in new stack
Oct 18 13:52:12 VERBOSE[12842] logger.c: -- Called
[EMAIL PROTECTED]/16079
99
Oct 18 13:52:12 VERBOSE[12842] logger.c: -- Call accepted by
208.139.204.245
(format ulaw)
Oct 18 13:52:12 VERBOSE[12842] logger.c: -- Format for call is ulaw
Oct 18 13:52:19 VERBOSE[12842] logger.c: -- IAX2/teliax-1 is making
progress
passing it to SIP/PLXFax-ceb5
Oct 18 13:52:19 DEBUG[12842] chan_iax2.c: Ooh, voice format changed to 4
Oct 18 13:52:37 DEBUG[12842] chan_iax2.c: Immediately destroying 1,
having recei
ved hangup
Oct 18 13:52:37 DEBUG[12842] chan_iax2.c: We're hanging up IAX2/teliax-1
now...
Oct 18 13:52:37 DEBUG[12842] chan_iax2.c: Really destroying
IAX2/teliax-1 now...
Oct 18 13:52:37 VERBOSE[12842] logger.c: -- Hungup 'IAX2/teliax-1'
Oct 18 13:52:37 VERBOSE[12842] logger.c: == No one is available to
answer at t
his time (1:0/0/0)
Oct 18 13:52:37 DEBUG[12842] app_dial.c: Exiting with DIALSTATUS=NOANSWER.
Oct 18 13:52:37 DEBUG[12842] cdr_addon_mysql.c: cdr_mysql: inserting a
CDR recor
d.
Oct 18 13:52:37 DEBUG[12842] cdr_addon_mysql.c: cdr_mysql: SQL command
as follow
s: INSERT INTO cdr
(calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,l
astdata,duration,billsec,disposition,amaflags,accountcode) VALUES
('2005-10-18 1
3:52:12','8667594678','8667594678','999','default',
'SIP/PLXFax-ceb5','IAX2/
teliax-1','Dial','IAX2/[EMAIL PROTECTED]/1607999|45|Tr',25,0,'NO
ANSWER',3,'')
Oct 18 13:52:37 DEBUG[12842] chan_sip.c: update_user_counter(PLXFax) -
decrement
inUse counter
Oct 18 13:52:37 DEBUG[12842] chan_sip.c: Stopping retransmission on
'e0dae938-16
[EMAIL PROTECTED]' of Response 102: Match Found
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Re: [Asterisk-Users] strange behavior after turning jitter buffer on
To avoid any confusion, you may note that the Dial Application does not time out in this log excerpt as I described. That's because I hung up the cell phone instead of waiting for the timeout. And before anyone asks, setting jitterbuffer=off made the problem go away. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: PRI echo issues: solvable?
On Tuesday 18 October 2005 12:18, Doug Meredith wrote: Andrew Kohlsmith [EMAIL PROTECTED] wrote: I've never seen that, it's always when we call out. Certain numbers will always trigger it. 888-737-4787 (IPC Resistors, it dumps into an IVR so it's safe to call) is one such number, but we have local numbers that hit other I just tried this number, and it was answered by a person. It's IVR most of time time. :-) Did you hear echo? -A. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Talkoff (Spurious DTMF) with 1.0.9.2 and TE406P
On Oct 18, 2005, at 2:03 AM, George Pajari wrote: We recently migrated a couple of PRIs to Asterisk 1.0.9.2 and a TE406P and are getting reports of talkoff (spurious/random DTMF tones heard by people on SIP equipment connected to the Asterisk server. We previously were using 1.0.3 with a T100P without any talkoff. (a) We have not set relaxdtmf. (b) There is no apparent pattern to the problem (seems to affect users on ATAs as well as SIP phones from different manufacturers). Any suggestions? Digium support can also help you with this, but as you load the module, there is a parameter that you can adjust the relaxedness of DTMF detection. It is called dtmfthreshold and it defaults to 1000. The higher that it is set to makes it more stringent in DTMF detection (i.e., less likely to have talk off). At about 2000-2500, most people stop getting DTMF events altogether, so that might be the range that you want to play around with. --- Matthew Fredrickson ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Forwarding Extensions using dialplan
Title: Forwarding Extensions using dialplan Hi all. So far this list is proving it's worth, even on my first day using it! I hope that someone might know an easy solution to this one. I would like to create a dial plan which will allow me to have all extensions 6XXX cause a dial-out of my T1 interface to a local number, wait for an answer, wait 2 seconds and then enter the extension. Can I do this in a dial plan somehow? This will allow me to pseudo-integrate a legacy telephone switch (whose extensions are all 6XXX) to my Asterisk system for direct extension dialing. David A. Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodata.net Tel: (519) 951-6079 Fax: (519) 451-6615 Poor planning on your part does not necessarily constitute an emergency on my part! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] zaptel problem
Hello people, i have a question concerning a quad-pri card (tor2 is the module for this card) i want the span to be completely shut down when alarms occur on it; i want the span to be shut down immediately to avoid compromising the whole box if one E1 line goes crazy and to be activated only by the administrator of the box... is there any posibility to do this? please note that i'm not a programmer and i don't know how to do this in C/C++ thank you, Calin S. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: PRI echo issues: solvable?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of alan
Sent: Tuesday, October 18, 2005 10:34 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Re: PRI echo issues: solvable?
Subject: RE: [Asterisk-Users] PRI echo issues: solvable?
Kris Boutilier [EMAIL PROTECTED] wrote:
On Tuesday 11 October 2005 11:49, alan wrote:
After solving the other low hanging fruit audio issues in our Asterisk
PBX, we are left with occasional cases of severe echo which we have not
found a solution for yet.
{clip}
Most calls have minimal, acceptable echo levels. But occasionally, we
get a call where the echo is delayed by a substantial amount (sometimes
around 250ms), and sounds as loud as the remote party.
{clip}
If you were to use ztmonitor on the channel to record the transmit and
receive sides to separate wav files, drive an impulse down the channel
(ie. a sharp, loud click) and then load the files into a tool where
they could be viewed side by side you'd see the actual echo endpath
(tail) length.
How does one use ztmonitor to record into separate files for transmit
and receive? My ztmonitor man page doesn't describe how to do this, it
only allows one -f File specification.
Oops. It seems I was thinking of cmd_monitor(), which records tx and rx legs
seperately... my error, sorry. Sounds like a good idea for a patch to
ztmonitor. :-)
When I monitored a sample conversation with ztmonitor, it
recorded both channels in one file. Then I set up a call from the number
which gives
us big echo. Although I heard echo when I was on the call, the
recorded version of the call did not record the echo.
There are two possibilities here:
1. it wasn't recording the incoming call leg.
2. the echo is entirely internal to our system
Almost there - I would suggest the echo is indeed present, but the time taken
for the echo to arrive back on the zap interface is imperceptibly small (ie
20ms) so you can't perceive it as an 'echo'. You might still be able to see the
reflection if the impulse is sharp (ie. short) enough by loading the wav file
from ztmonitor into Sonogram (http://www.dfki.de/~clauer/sonogram/) and
visually examining the waveform. If the echo delay path is too short to see
then the reflection will have been merged with the transmitted signal in the
consolidated file and have just resulted in an increase in amplitude.
It's the time delay due to processing and conveyance inside Asterisk and
everything else between your endpoint and the zap interface that causes the
reflection to change from 'sidetone' to 'perceptible echo'. cmd_monitor()
should be recording after at least some amount of delays are introduced, thus
the echo should be clearly audible there.
It's very important to understand that short echo paths (ie. 20ms) occur
quite frequently in the PSTN but unless something introduces an additional
delay into the signal path, and ISDN based digital PBXs such as the Norstar or
Meridian don't, the echo can't be perceived. Thus Asterisk, because everything
it does involves packetization and it's associated processing and conveyance
delays, needs to meet a much higher standard of echo cancellation.
Hope that helps.
Kris Boutilier
Information Services Coordinator
Sunshine Coast Regional District
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RE:[Asterisk-Users] free dids on goiax.com
I have been using goiax for my outgoing and has been excellent in quality ofvoice and also the service.To put the abusers out, I propose that each account holder shouldpre-register the phone numbers they typically call. For a legitimate user,pre-registration is certainly acceptable (especially for a free service),and if we give option of uploading an outlook address book or so, that wouldeven simplify the process. Pre-registration of outgoing calls will makeabusers' job lot harder and it is lot easier for hunting down the abusers aswell. If you want, you can set a maximum pre-registration limit as well.To avoid multiple account holders, i would take an approach of comparing IPaddresses at the time of registration itself. Suppose, if I want to create100 accounts for 100 email accounts, i will have to go to goiax.com andregister each and everyone of them; then collect the IP address of therequest and check against the existing accounts; you can set a maximum capon the number of accounts that could be registered from an IP (to allowroommates). If you set a max-cap to be 4, we can have all 4 roommates havetheir own accounts with each having pre-registered phone numbers. You canuse similar tactic to restrict the service users as well at the time ofallowing the outgoing call... just my thoughts to get the great service going.regards,Rajesh- Original Message - From: "Matthew Simpson" [EMAIL PROTECTED]To: asterisk-users@lists.digium.comSent: Tuesday, October 18, 2005 2:05 AMSubject: [Asterisk-Users] free dids on goiax.com GoIAX, the Asterisk community's free IAX provider, is offering free US dids now. I loaded about 175 dids in and put up a very beta sign in page. Unfortunately I had to restrict the free us/canada outbound calling back down to toll-free only. There was a lot of war dialing and prank calling going on. I'm working on some stuff to hopefully curb that kind of stuff down so I can unrestrict outdial again, but this is the problem with free.. there is always someone that will abuse it. If anybody has any ideas on how to keep the abuse down let me know. The best ideas I have now is to only allow a certain amount of calling per month, add velocity checking, and somehow put some accountability into the sign up process to keep the prank callers and multiple account abusers away. yours, Matthew Simpson GoIAX -- www.goiax.com TxLink -- www.txlink.net ___ --Bandwidth and Colocation sponsored by Easynews.com -- ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dial Plan
Hi, I´ve just installed an Asterisk Server on a Fedora Core 4, and made it work between diferrent extensions in the office and now I need to make it work on calling outside the office and I think I need a Dial Plan, can somebody help me a little with this? Thanks a lot ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] One phone ringing, one phone flashing ?
Hi, well, some clients have strange ideas and wishes (at least to my mind). Yesterday I gave a presentation about asterisk to a CEO. At the end he asked me whether asterisk is able to do the following: When a call for the CEO comes in, the calling number should be shown on the display of his phone and the phone of his secretary. The secretary's phones should ring, but at his phone only a light should flash. ;-)) No, turning off the sound isn't the solution. This restriction should e.g. only apply, when it is an external call, internal calls should result in ringing both phones. I'm not quite sure, whether this could be a feature of asterisk or the phone or both together. Does anything of you successfully set up something like this or could recommend a phone that would help/support it? Thanks a lot in advance, Stefan ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Forwarding Extensions using dialplan
On Tue, 2005-10-18 at 14:50 -0400, Dave Morrow wrote:
Hi all. So far this list is proving it's worth, even on my first day
using it!
I hope that someone might know an easy solution to this one.
I would like to create a dial plan which will allow me to have all
extensions 6XXX cause a dial-out of my T1 interface to a local number,
wait for an answer, wait 2 seconds and then enter the extension. Can
I do this in a dial plan somehow? This will allow me to
pseudo-integrate a legacy telephone switch (whose extensions are all
6XXX) to my Asterisk system for direct extension dialing.
It can, however how difficult this is varies greatly on the mapping
between old and new 6xxx numbers.
If you are using 1.1 or 1.2 code you can do this:
exten = _6xxx,1,Dial(Zap/1/${EXTEN},60,M(dialexten^${EXTEN}))
[macro-dialexten]
exten = s,1,wait(2)
exten = senddtmf(${arg1})
This causes asterisk to pattern match 6xxx where x is 0-9. When the
called party answers the macro dialexten is called and that will send
the dtmf of the extension in question. If the two are the same then no
translation is needed, if they are different then you have to play games
to match the new with the old.
--
Trixter http://www.0xdecafbad.com Bret McDanel
UK +44 870 340 4605 Germany +49 801 777 555 3402
US +1 360 207 0479 or +1 516 687 5200
FreeWorldDialup: 635378
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Re: [Asterisk-Users] huge problem compiling * on gcc4.x (SUSE 10.0)
On Tue, Oct 18, 2005 at 09:10:38AM +0200, Tzafrir Cohen said: On Mon, Oct 17, 2005 at 07:01:17PM -0400, Walt Reed wrote: I was unable to get a clean compile of the kernel or * with gcc 4. You can ask about this in Debian lists. I don't have unstable so I can't test for myself, but the current unstable kernel surely builds for them. Sure, but it depends on which compiler they use... They may not use 4 on the kernel. In fact, I'm pretty sure they don't. The version of the kernel I was compiling is the version WITH debian patches (latest.) And it surely does not compile with gcc 4. As for Asterisk 1.2: It should hit experimental any day now. There are also unofficial debs at http://rapid.dotsrc.org/experimental/ . If those don't build with gcc 4 then this should be reported. Most gcc 4 incompatibility bugs I saw were fixed pretty fast. The incompatabilities are going to be kernel compiled with one version of GCC, zaptel module compiled with another. That, even if it appears to work, is not a good idea. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Fax2Mail
I don't know of a comprehensive guide, but you can set it up using
NVFaxDetect, NVFaxEmail, and SpanDSP or Hylafax. NVFaxEmail can pull the
e-mail addresses from it's own config, voicemail.conf, a database, or thru
realtime.
Simple extensions.conf:
[incoming-dids]
exten = _541359,1,NVFaxDetect(...) ; Make sure this is a fax
exten = fax,1,NVFaxEmail(...,${CALLERID},pu,...) ; Receive and e-mail PDF
with user lookup
-J
--
Message: 10
Date: Tue, 18 Oct 2005 07:39:10 -0700 (PDT)
From: David [EMAIL PROTECTED]
Subject: [Asterisk-Users] Fax2Mail
Hello,
Is there or can anyone provide a comprehensive guide (designed for
Linux/Asterisk novices) to installing/setting up Asterisk in order to
support Fax2Mail service?
In my case, I would like Asterisk to receive fax calls to predefined
numbers (ranges) and to associate each of these numbers to email addresses.
Thank you in advance.
David
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RE: [Asterisk-Users] Fwd: {100-1287} RE: DIDs
Why don't you just try calling sellvoip directly? They are very responsive via
phone and email normally...
Their numbers are right on the website.
John
-Ursprüngliche Nachricht-
Von: Paul [mailto:[EMAIL PROTECTED]
Gesendet: Tuesday, October 18, 2005 9:55 AM
An: Asterisk Users Mailing List - Non-Commercial Discussion
Betreff: Re: [Asterisk-Users] Fwd: {100-1287} RE: DIDs
Jerry if you have something to ask or say about your vendor on this list
do so.
But please stop dumping a copy here of all communications with them.
Jerry Richmond wrote:
Note: forwarded message attached.
big snip
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Re: [Asterisk-Users] One phone ringing, one phone flashing ?
Message: 18Date: Tue, 18 Oct 2005 21:02:28 +0200From: Stefan-Michael. Guenther (in-put GbR) [EMAIL PROTECTED]Subject: [Asterisk-Users] One phone ringing, one phone flashing ?To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED]Content-Type: text/plain;charset=utf-8Hi,well, some clients have strange ideas and wishes (at least to my mind). Yesterday I gave a presentation about asterisk to a CEO.At the end he asked me whether asterisk is able to do the following:When a call for the CEO comes in, the calling number should be shown on the display of his phone and the phone of his secretary. The secretary's phonesshould ring, but at his phone only a light should flash.;-)) No, turning off the sound isn't the solution.This restriction should e.g. only apply, when it is an external call, internalcalls should result in ringing both phones.I'm not quite sure, whether this could be a feature of asterisk or the phoneor both together.Does anything of you successfully set up something like this or could recommend a phone that would help/support it?Thanks a lot in advance,Stefan The quick and dirty solution to this that I'd put together would involve a Cisco or Polycom multiline phone, and some careful magic with ring tones. First, set up two different lines on the boss's phone- one that you'll use for inbound external calls, the other that you'd use for inbound internal calls. When a new call came in, the call would be routed simultaneously (through DIAL(TECH/EXTNTECH/EXTN2) ) to both the secretary's phone and to the boss's phone external line. The ringer can be independantly specified on each line for the Cisco phone- so you specify a custom ringtone that's essentially silent- record some dead air for his inbound external line. Since the Cisco displays CID and flashes the red light on ringing, he gets his flashing light- and it rings as needed at the secretaries desk. Cheap, dirty solution- but functional in a short timeframe. -pbd. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IP300 - Asterisk - Broadvoice - PSTN Choppy / cuts in and out
Hello All - I've got an asterisksetup using 3 broadvoice lines and 5 Polycom IP300 phones. We have 1.5Mbit up and down via cable. 40ms (ave) pings to the broadvoice proxy and no packetloss. The phones sound like cell phones. The person on the other end complains about it cutting in and out. On our end, it cuts in and out as well. Within the office, we can call from one IP300 to another with absolutely no problems at all. Sounds great. We are connected through a Linksys (Firmware v1.05.0). Wired QoS is enabled with the asterisk box's mac being highest priority, and everything else being low. Upstream bandwidth is set to Auto. [I doubt these settings are the problem as the choppy/cell-phone-sounding effect also occurs when there is minimal network traffic.] Any help troubleshooting this plz? Snippets fromsip.conf: [210]username=210type=friendsecret=***record_out=On-Demandrecord_in=On-Demandqualify=noport=5060nat=never[EMAIL PROTECTED]host=dynamic dtmfmode=rfc2833context=from-internal-bbpcanreinvite=nocallerid=Mike 210 [bbpbv1]username=949743user=phonetype=peersecret=**nat=yesinsecure=veryhost=sip.broadvoice.comfromuser=949743fromdomain= sip.broadvoice.comdtmfmode=inbandcontext=from-bbp-pstncanreinvite=noauthname=949743 [949743]username=949743user=949743type=usernat=yesinsecure=veryhost=sip.broadvoice.comfromdomain=sip.broadvoice.com dtmfmode=inbanddtmf=inbandcontext=from-bbp-pstn Test call: # asterisk -vrx sip show channelsPeer User/ANR Call ID Seq (Tx/Rx) Format147.135.8.128 1949# 5855b260439 00103/0 ulaw 192.168.1.100 210 f8c9ee5e-9f 00101/2 ulaw 2 active SIP channel(s) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX only speech one way
Hello I have two Asterisk's connected via IAX, they are sitting on the same network, via a VPN, so there should be no problems with firewalls. My problem is that when a person calls from A to B, A will not hear B speak. B hears A fine. I doesn't matter who initiates the call. One of the Asterisk'ses is a new installation, just installed, but with the Conf-files from an earlier setup, that worked fine. Asterisk version on computer A is Asterisk CVS-v1-0-12/09/04-08:58:31 Asterisk version on computer B is Asterisk CVS-D2005.05.28.22.00.00-10/17/05 Two different versions, but I dont think it should matter? Michael ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dial Plan
Felix Amaral wrote:
Hi, I´ve just installed an Asterisk Server on a Fedora Core 4, and made it
work between diferrent extensions in the office and now I need to make it
work on calling outside the office and I think I need a Dial Plan, can
somebody help me a little with this?
I have the following at the bottom of my context for my SIP
extensions:
exten = _1.,1,Dial(SIP/pstn/${EXTEN})
; Unanswered, Do we ever get here?
exten = _1.,2,Playback(allison7/all-circuits-busy-now)
exten = _1.,3,Playback(allison7/pls-try-call-later)
exten = _1.,4,Macro(hangupcall)
; Busy, Or do we always go here?
exten = _1.,102,Playback(allison7/pls-try-call-later)
exten = _1.,103,Playback(allison7/all-circuits-busy-now)
exten = _1.,104,Macro(hangupcall)
I only have a Grandstream BT100 a Sipura SPA-3000 (named line1
pstn). The SPA-3000 registers pstn with asterisk and is able to
handle all the dial outs. Since I no longer have 7 digit dialing
(we have overlays which require dialing at least 10 digits). So
any number that starts with 1 will be sent out to the PSTN. I
haven't setup the emergency services number or other numbers
such as emergency dialing. I really aught to do that in case
someone accidentally picks up my extra test phones.
BTW, this is for home use.
--
Linux Home Automation Neil Cherry [EMAIL PROTECTED]
http://home.comcast.net/~ncherry/ (Text only)
http://hcs.sourceforge.net/ (HCS II)
http://linuxha.blogspot.com/My HA Blog
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RE: [Asterisk-Users] free dids on goiax.com
Why not just ask for a small one time payment $1 or something from a credit card, or paypal, or something along those lines so you would have someway to trace back to an abuser. John -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Simpson Sent: Tuesday, October 18, 2005 3:05 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] free dids on goiax.com GoIAX, the Asterisk community's free IAX provider, is offering free US dids now. I loaded about 175 dids in and put up a very beta sign in page. Unfortunately I had to restrict the free us/canada outbound calling back down to toll-free only. There was a lot of war dialing and prank calling going on. I'm working on some stuff to hopefully curb that kind of stuff down so I can unrestrict outdial again, but this is the problem with free.. there is always someone that will abuse it. If anybody has any ideas on how to keep the abuse down let me know. The best ideas I have now is to only allow a certain amount of calling per month, add velocity checking, and somehow put some accountability into the sign up process to keep the prank callers and multiple account abusers away. yours, Matthew Simpson GoIAX -- www.goiax.com TxLink -- www.txlink.net ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Dial Plan
The Asterisk I biult only does outbound calls, and it do them by LAN, I
don´t have any special hardware. Please help with the Dial Plan.
Thanks a lot
Felix Amaral
I.T. - Information Technology
Grupo PyD S.A.
Reconquista 1011 4º (C1003ABU)
Cap. Fed.- Argentina
TeL: +54-11--4800 Ext. 555
[EMAIL PROTECTED]
http://www.grupopyd.com
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Neil Cherry
Enviado el: Martes, 18 de Octubre de 2005 05:17 p.m.
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Asunto: Re: [Asterisk-Users] Dial Plan
Felix Amaral wrote:
Hi, I´ve just installed an Asterisk Server on a Fedora Core 4, and
made it work between diferrent extensions in the office and now I need
to make it work on calling outside the office and I think I need a
Dial Plan, can somebody help me a little with this?
I have the following at the bottom of my context for my SIP
extensions:
exten = _1.,1,Dial(SIP/pstn/${EXTEN}) ; Unanswered, Do we ever get here?
exten = _1.,2,Playback(allison7/all-circuits-busy-now)
exten = _1.,3,Playback(allison7/pls-try-call-later)
exten = _1.,4,Macro(hangupcall)
; Busy, Or do we always go here?
exten = _1.,102,Playback(allison7/pls-try-call-later)
exten = _1.,103,Playback(allison7/all-circuits-busy-now)
exten = _1.,104,Macro(hangupcall)
I only have a Grandstream BT100 a Sipura SPA-3000 (named line1 pstn).
The SPA-3000 registers pstn with asterisk and is able to handle all the dial
outs. Since I no longer have 7 digit dialing (we have overlays which require
dialing at least 10 digits). So any number that starts with 1 will be sent
out to the PSTN. I haven't setup the emergency services number or other
numbers such as emergency dialing. I really aught to do that in case someone
accidentally picks up my extra test phones.
BTW, this is for home use.
--
Linux Home Automation Neil Cherry [EMAIL PROTECTED]
http://home.comcast.net/~ncherry/ (Text only)
http://hcs.sourceforge.net/ (HCS II)
http://linuxha.blogspot.com/My HA Blog
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[Asterisk-Users] Monit test for IAX2
Has anyone got a monit test written for IAX2? I've tried: check host blah with address blah if failed port 4569 use type udp then alert But it seems to pass even when I choose a fake port that I know is not open, like 4500 I'm wondering if someone has used send|expect to do a basic IAX2 protocol test? thx - Alex ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: RE:[Asterisk-Users] free dids on goiax.com
On Tue, 2005-10-18 at 13:52 -0500, Rajesh kumar wrote: To put the abusers out, I propose that each account holder should pre-register the phone numbers they typically call. For a legitimate user, pre-registration is certainly acceptable (especially for a free service), and if we give option of uploading an outlook address book or so, that would even simplify the process. Pre-registration of outgoing calls will make abusers' job lot harder and it is lot easier for hunting down the abusers as well. If you want, you can set a maximum pre-registration limit as well. Making lists of who you are going to call defeats the purpose. If I only wanted to call a small number of people I wouldnt need US48 termination. If I have to upload a contact list (or enter it manually which would be a lot more bothersome) I wouldnt use the service. The reason for that is I dont know who I am gonna call, and I dont give out my address book to some website just because. I would imagine that a lot of other people would feel the same. Now lets say a new pizza place opens up and runs a local commercial on TV. Some number if flashed on the screen but its not in my address book. I only use goiax for outbound. I then have to register that number to call to order a pizza? Not very user friendly. A friend is staying in a hotel and I wish to call them I have to add that hotel as well? To avoid multiple account holders, i would take an approach of comparing IP addresses at the time of registration itself. What about all the proxy lists that exist freely available? What about proxy scanners that exist to find new unlisted ones? What about the fact that with most cable modem providers its trivial to get a new IP. Then there is the issue of dynamic IPs where people might sign up then get a new IP someone new comes along and cant sign up becuase that IP is already flagged as being at its maximum. So you would have to timestamp and purge the database occasionally. A lot of overhead for a system so easily defeated. While I appreciate the problems Matthew is going through, this is a complex issue, and one that has plagued the net for a long time. How do you authenticate random people on the internet as 1. unique and 2. as themselves. The net provides anonymity and without associating a physical mail address and mailing them a code (slow, costly, etc) its really hard to do that for a free service. Credit cards can be used to a degree to do this, but for a free service I seriously doubt anyone will enter a credit card. And on the same token I doubt that anyone would scan in and email a copy of their drivers license either. If someone comes up with a way to authenticate users with no prior contact that users will accept and adhere to for a free service like this they could make a ton of money overnight because that is kinda one of the holy grails of authentication that is desired right now (as it has direct impacts on paid services). This is a very complex problem and so far the best methods require other forms of authentication based on preexisting ones (ie credit card verification to match against) or are costly (a code mailed to the person). -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
