Re: [asterisk-users] weird undocumented extensions such as s-BUSY
Am Dienstag, den 23.01.2007, 05:41 -0200 schrieb Barzilai Spinak:
I've seen several examples that use extensions such as;
s-BUSY
s-NOANSWER
etc...
It's more or less evident what they do, but I've searched for some
FORMAL documentation everywhere and have found nothing.
Do they work for anything else than s-? (I think I've seen other
examples, but can't find them now)
Are they standard in any way?
What are the allowed values after the dash?
In which version were they introduced?
etc...
Those are not standardized extensions. They will be called _ONLY_ by a
Goto() command, like in
exten = s,1,Dial(SIP/user1,60,tT)
exten = s,2,Goto(s-${DIALSTATUS})
so you can choose to put names as you like them before the dash, the
DIALSTATUS variable values will be explained in
http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial and
http://www.voip-info.org/wiki/index.php?page=Asterisk+variable
+DIALSTATUS
I personally prefer to do that with GoTos inside the extension, or using
Macros, but that is up to everyone's personal choice.
BR
Anselm
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RE: [asterisk-users] X100P how do i recieve incomming calls?
There is only one x100p card in the system thanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: 22 January 2007 23:40 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] X100P how do i recieve incomming calls? On Mon, Jan 22, 2007 at 08:08:16PM -, Charlie Grosvenor wrote: I have just purchased a 2nd hand X100P, Is there another X100P card in the same system? if I do a ztcfg -vv I get: Zaptel Version: 1.4.0 Echo Canceller: MG2 Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) 1 channels configured. My understanding of the above is that the zaptel driver has detected the card. What do I now need to do, in order to get an incoming call to work with asterisk? I assume I need to make some sort of change to /etc/asterisk/zapata.conf in order to tell asterisk about the card? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] STUN and SNMP
22 jan 2007 kl. 07.38 skrev Thomas Deillon: Hi all, I read somewhere that asterisk v 1.4 can make Stun and SNMP. I tried to find more information on these features but I didn’t find any clues. Someone find a way to use it? There's a module called res_snmp that implements an SNMP agent or an NetSNMP agent plugin. You need to have Netsnmp installed for this to be compiled, as well as have it enabled in menuselect. The stun support is only implemented in the google talk/jingle channel driver. /O___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Requirements for faxes to work properly
22 jan 2007 kl. 16.13 skrev dima: Hello, everyone. I'm reading about the asterisk new features. One is T.38 protocol support. I used faxes before with asterisk 1.2 and everything was working quite well. Could anyone explain what have changed in the way faxes are handled. Another thing is, in order for asterisk to work over T.38 with my fax machine do I also need a T.38 support from my ATA and my SIP provider? The T.38 support in Asterisk 1.4 is only pass through. Asterisk is not an endpoint or gateway of any sort. It only works passthrough in SIP to SIP calls without chan_local or chan_agent involved. Both endpoints of the call will need T.38 support and have T.38 enabled in sip.conf. Asterisk 1.4 as I see it is still a bit unstable and not ready for production use. Be careful out there. /O ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Problems with rxfax
Just out of curiosity. Would you mind sharing that app_rxfax.c file that you modified to work with SpanDSP 0.0.3? I wouldn't mind, I've attached the sources of both rx- and txfax here, I hope you can use them. About IAXmodem with hylafax, we've also tried that but we ran into the same problems, a lot of the received faxes were missing large parts. We chose to continue with app_rxfax since it is easier to configure and easier to hack ;) and it doesn't add an extra VOIP channel to the chain. Since both IAXmodem and app_rxfax use spandsp we figured that we do something wrong in the way we use spandsp, but we have no idea what that might be or how we can check this. I hope that somebody has some additional information? Thanks in advance, Ardjan Zwartjes, Telecats. app_txfax.c Description: app_txfax.c app_rxfax.c Description: app_rxfax.c ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Detecting Disconnected Numbers - PRI
The correct way to determine the ending cause of a call is the
${HANGUPCAUSE} variable that Dial creats. Just to be sure, set
priindication=outofband in /etc/asterisk/zapata.conf. HANGUPCAUSE
should always be set.
HANGUPCAUSE is indeed always set. The question is, Set with what data?
The problem is that the telco doesn't consistently and uniformly send
back the Q.931 hangup cause. Believe me, I've pored over mountains of
Q.931 logs, both with inband and outofband signaling. The telcos just
plain suck at delivering this information consistently. They usually
get it right, but when you are making tens of thousands of dial attempts
per day and the telco is giving you accurate info 90% of the time then
you still have 100's of call records with suspect data. Garbage in,
garbage out.
My work around is to make multiple attempts on so-called invalid numbers
and to keep track of the results. If I dial a phone number and get
hangup cause 16 less than two seconds after the dial attempt, and if I
can repeat that result, then I assume it is truly a disconnected or
otherwise invalid number.
-MC
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[asterisk-users] Operate on registrations
Hi, I have a bunch of SIP phones(behind NAT) registering on my * box. I want to find out when they register and de-register. I also want to operate on it, so when they register/de-register, I want to insert calldate into a mysql DB, etc. Maybe this will help me when, for instance a user tries to register but has the wrong username/password. Now I am aware of regcontext, but it only creates a 1,NoOP for that user, I want it to execute that, so I can have this maybe: exten = 666,2,AGI(Registraion.agi) so when my users register 666,1,NoOp will be created and execution can start there. Any Ideas on how I can get something like this? -- thanks, Yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] weird undocumented extensions such as s-BUSY
Barzilai Spinak wrote:
I've seen several examples that use extensions such as;
s-BUSY
s-NOANSWER
etc...
It's more or less evident what they do, but I've searched for some
FORMAL documentation everywhere and have found nothing.
Do they work for anything else than s-? (I think I've seen other
examples, but can't find them now)
Yes.
Are they standard in any way?
I'd say no, but it depends how you look at it.
What are the allowed values after the dash?
Any potential value of ${DIALSTATUS}
In which version were they introduced?
Not sure, but they are just various destinations for a jump caused by
Goto(s-${DIALSTATUS})
asterisk doesn't know about and doesn't care about these extensions;
there is nothing special about them and that's why you can't find any
documentation.
HTH,
Trevor
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Re: [asterisk-users] No Audio for Extension to Extension
On 22 Jan 2007, at 07:28, Troy - Purple Oranges wrote: I am at a loss, I can terminate and receive calls via any of my providers with both IAX and SIP. I use GSM, G729a, and ulaw for those carriers. If I make an extension to extension call - there is no audio at all in either direction. All my extensions are set to use G729a (I have tried changing that though to see if it would fix it). I am fairly sure it is not a transcoding issue - as the server transcodes for the inbound/outbound calls. You really need to tell us more! At a pure guess however I'd say you have SIP extensions with canreinvite set to true. Your internal network however does not permit rtp traffic between the handsets. Tim Panton www.mexuar.net www.westhawk.co.uk/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 7 points of comparison Polycom 430/501 and Aastra 480i. Which one to choose ?
On 1/22/07, Vikas [EMAIL PROTECTED] wrote: I need to provide a 80 people office with VOIP. I want to commit to one vendor Polycom or Aastra. Price of the phones is not a factor in the decision. The quality of the phones is the factor. Some of the features that I am evaluating on are: (arranged in order of priority) 1. Sound quality 2. complete product line with conference phone and receptionist phone (not on Aastra) 3. cordless (not on 501/430) 4. backlit LCD (not on 501/430) 5. Inbuilt POE (not on 501) 6. speaker phone 7. 2 network ports. Which one will you choose ? Get a couple of each phone and live with it for a week. You will soon find which one you like. Then repeat the process with a couple of 'decision makers'. Helpful though this list is, it can't take into account local factors like desk size, nature of business, office environment etc all of which play into the choice of phone. Perhaps you should be looking at the SNOM too ? Tim Panton www.mexuar.net www.westhawk.co.uk/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Can't find asterisk.ctl under CentOS installation
Hi Everyone, I recently upgraded to Asterisk 1.4 using the RPMS at ATrpms.net on CentOS 4.4, Asterisk starts up but when I start the console it reports this error and drops out. Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist)? I have checked to see that the file asterisk.ctl actually exists. Any suggestions? -- I never look back darling, it distracts from the now, Edna Mode (The Incredibles) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dial plan constructions suggestions?
Can I ask for some advice on dial-plan construction please I have setup my dialplan to use 9 to get a zap trunk, leaving everything else for internal extensions. However, this creates a problem in that my callerid is correct, but doesn't work to re-dial the incoming caller. So if I simply click missed calls on my Snom phone and hit redial then it tries to dial an internal extension. So I then setup Asterisk to add a 9 to the incoming callerid for all calls which come via the Zap trunk, but now this creates some issues with applications like Snapanumber and perhaps HudLite, which are trying to map the caller ID to numbers in the addressbook (and I don't really want my internal Outlook address books to have everyone listed with a 9 in front of their number) How are others handling this? I have considered simply dropping the prefix digit and working around any clashes in internal and external numbers (not very hard). Grateful for any thoughts Ed W ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to exit from console?
Hi, all Stupid question, but how do you exit asterisk console without stopping the asterisk? Tried quit and exit: *CLI exit No such command 'exit' (type 'help' for help) *CLI quit No such command 'quit' (type 'help' for help) *CLI Any other ideas? I started asterisk with -cg option. Same problem if use asterisk -r to connect. Can not exit. Any ideas? Thanks, Rudolf ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] beronet BRI card sometimes not detecting tones
Hi, I have an asterisk 1.2.9.1 box with a beronet card (install-misdn-mqueue). It is working good but some callers can call the number to my PBX, hears the intro message but cannot choose which extension to call (es: 1,2,...,104, etc) because the card does not accept extra digits. Is there anybody who knows why and how to solve the prob? TIA Giorgio ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to exit from console?
Try safe_asterisk , for an easy way to start asterisk in background, and then connect with asterisk process running asterisk -rx Now you can use exit, and by the way you may look on wiki diferent ways to run asterisk. On 1/23/07, Rudolf Ladyzhenskii [EMAIL PROTECTED] wrote: Hi, all Stupid question, but how do you exit asterisk console without stopping the asterisk? Tried quit and exit: *CLI exit No such command 'exit' (type 'help' for help) *CLI quit No such command 'quit' (type 'help' for help) *CLI Any other ideas? I started asterisk with -cg option. Same problem if use asterisk -r to connect. Can not exit. Any ideas? Thanks, Rudolf ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to exit from console?
Am 23.01.2007 um 12:36 schrieb Rudolf Ladyzhenskii: Stupid question, but how do you exit asterisk console without stopping the asterisk? Tried quit and exit: *CLI exit No such command 'exit' (type 'help' for help) *CLI quit No such command 'quit' (type 'help' for help) *CLI Any other ideas? I started asterisk with -cg option. Same problem if use asterisk -r to connect. Can not exit. If you startet with -c Asterisk doesn't allow you to exit. You have to stop now: ---cut--- *CLI exit No such command 'exit' (type 'help' for help) *CLI stop now debian:~# ---cut--- Any ideas? Start Asterisk normal with a simple asterisk and than log into the CLI with asterisk -r THAN you can use exit. Stefan -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to exit from console?
On Tue, 23 Jan 2007 22:36:12 +1100 Rudolf Ladyzhenskii [EMAIL PROTECTED] wrote: Hi, all Stupid question, but how do you exit asterisk console without stopping the asterisk? Tried qui Any other ideas? I started asterisk with -cg option. Same problem if use asterisk -r to connect. Can not exit. Run asterisk just by typing 'asterisk'. Using the -c option will cause the behaviour you are experiencing. Then when you connect with 'asterisk -r', you can use 'exit' (or just ctrl-c) to disconnect, but leave Asterisk running in the background. On the console, you can get the - behaviour with 'set verbose 4' Cheers, Gavin. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dial plan constructions suggestions?
I don't know about SNOM, but with Xlite Softphone you can have the SoftPhone internal dialplan. Ex. [29];match=1;pre=0; this adds a Zero to every nine digits number s I dial begining with 2 or 9 , this has nothing to do with asterisk, is VoiP phone dialplan. So you can tell to the softphone that when you dial a specific pattern like 9 the Softphone should add an extra 9 in the beginning. This helped me out to import all my contacts from Outlook without having to Add a 0 in the begining of all of them. Hope this help On 1/23/07, Ed W [EMAIL PROTECTED] wrote: Can I ask for some advice on dial-plan construction please I have setup my dialplan to use 9 to get a zap trunk, leaving everything else for internal extensions. However, this creates a problem in that my callerid is correct, but doesn't work to re-dial the incoming caller. So if I simply click missed calls on my Snom phone and hit redial then it tries to dial an internal extension. So I then setup Asterisk to add a 9 to the incoming callerid for all calls which come via the Zap trunk, but now this creates some issues with applications like Snapanumber and perhaps HudLite, which are trying to map the caller ID to numbers in the addressbook (and I don't really want my internal Outlook address books to have everyone listed with a 9 in front of their number) How are others handling this? I have considered simply dropping the prefix digit and working around any clashes in internal and external numbers (not very hard). Grateful for any thoughts Ed W ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Detecting Disconnected Numbers - PRI
Mike, my friend, you have hit the nail on the head - and thanks for the
support - it's good to know I'm not alone with this issue.
I am working with 4 customer callcentre sites to resolve this problem. 3
sites are in Melbourne (Aus) and one in Auckland. The Auckland site is
dialing international back to Australia.
Oddly, the telco in New Zealand is providing a much richer PROGRESS
indication set for internationally dialled numbers than I am getting for the
same numbers dialled locally from here in Melbourne. Although it's not true
for all numbers, the ones in question have no cause code associated with the
PROGRESS indication and they all seem to have voice treatment during the
PROGRESS indication - kindly telling me that I have dialled a wrong number,
or that it is going to divert off somewhere else.
All sites have requested out of band indications for PRI, but it looks like
the only way to resolve this issue once and for all telcos is to assume that
we are going to hang up immediately we receive a PROGRESS indication. I know
this is not ideal and will result in quite a few false positives but it is
likely to be right for more numbers than it is going to be wrong.
I'd be grateful for your thoughts on this direction. Your comments so far
have been very useful.
As for the telco's saying PRI is good but not perfect - I would struggle
to understand how they would improve on this position. I mean what else is
available for multi-channel exchange termination?
cheers,
Mark.
On 1/23/07, Michael Collins [EMAIL PROTECTED] wrote:
The correct way to determine the ending cause of a call is the
${HANGUPCAUSE} variable that Dial creats. Just to be sure, set
priindication=outofband in /etc/asterisk/zapata.conf. HANGUPCAUSE
should always be set.
HANGUPCAUSE is indeed always set. The question is, Set with what data?
The problem is that the telco doesn't consistently and uniformly send
back the Q.931 hangup cause. Believe me, I've pored over mountains of
Q.931 logs, both with inband and outofband signaling. The telcos just
plain suck at delivering this information consistently. They usually
get it right, but when you are making tens of thousands of dial attempts
per day and the telco is giving you accurate info 90% of the time then
you still have 100's of call records with suspect data. Garbage in,
garbage out.
My work around is to make multiple attempts on so-called invalid numbers
and to keep track of the results. If I dial a phone number and get
hangup cause 16 less than two seconds after the dial attempt, and if I
can repeat that result, then I assume it is truly a disconnected or
otherwise invalid number.
-MC
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--
regards,
Mark P. Edwards
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Re: [asterisk-users] tdm400p not working with brazilian lines
Hi Tzafrir, the caller hear the right tone but when the calle picks up the phone Asterisk hangs up. The caller hears the call has answered and hung up in less than a second. We tried the reversepolarity parameter...it initially seemed to work but after some call had the same problem. We think it may depend from telco line but has no info about brazilian telcos...that's why I'm searching for some guy who knows brazilian analog lines. Giorgio Tzafrir Cohen wrote: On Mon, Jan 22, 2007 at 03:16:54PM +0100, Giorgio Incantalupo wrote: Hi, I'm installing an Asterisk box with a TDM2400P in Brazil. I can make analog phones work while lines are not working. What does happend when you try to ring or when a call comes in? Since I do not know anything about brazilian lines, is there anybody who can tell me what is wrong/missing in my conf files (below)? TIA Giorgio _zaptel.conf:_ fxoks=9-16 fxsks=17-24 defaultzone=br loadzone=br* * _zapata.conf:_ context = inbound_zap echocancel = 128 echocancelwhenbridged = yes echotraining = 200 language = br signalling = fxo_ks callerid = Christina 102 channel = 9-16 context = outbound_zap canpark = yes echocancel = 128 echocancelwhenbridged = yes echotraining = 200 faxdetect = both language = br musiconhold = native signalling = fxs_ks callerid = asreceived channel = 17-24 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to exit from console?
Stop now Em Terça 23 Janeiro 2007 09:41, Marco Mouta escreveu: Try safe_asterisk , for an easy way to start asterisk in background, and then connect with asterisk process running asterisk -rx Now you can use exit, and by the way you may look on wiki diferent ways to run asterisk. On 1/23/07, Rudolf Ladyzhenskii [EMAIL PROTECTED] wrote: Hi, all Stupid question, but how do you exit asterisk console without stopping the asterisk? Tried quit and exit: *CLI exit No such command 'exit' (type 'help' for help) *CLI quit No such command 'quit' (type 'help' for help) *CLI Any other ideas? I started asterisk with -cg option. Same problem if use asterisk -r to connect. Can not exit. Any ideas? Thanks, Rudolf ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Did you know that for the price of a 280-Z you can buy two Z-80's? -- P. J. Plauger ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to exit from console?
Rudolf Ladyzhenskii wrote: Hi, all Stupid question, but how do you exit asterisk console without stopping the asterisk? If you start Asterisk without any options: asterisk And then reconnect to it via the -r option asterisk -r Then typing exit on the console will exit without stopping Asterisk. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] No Mailbox Prompt
Hello list,
I have the following extension rule configured to transfer incoming calls to
Voicemail directly:
exten = _123105.,1,Voicemail(su${EXTEN:[EMAIL PROTECTED])
exten = _123105.,2,Hangup
Is it possible to add a Playback line before the hangup (or before the
voicemail) in case someone reaches an extension that doesn't have an active
mailbox? Something like:
exten = _123105.,2,Playback(no-box,noanswer)
Thanks.
David.
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Re: [asterisk-users] No Audio for Extension to Extension
enable rtp debug in your asterisk CLI and check if there's traffic passing. Would be a first approach I think. On 1/23/07, Tim Panton [EMAIL PROTECTED] wrote: On 22 Jan 2007, at 07:28, Troy - Purple Oranges wrote: I am at a loss, I can terminate and receive calls via any of my providers with both IAX and SIP. I use GSM, G729a, and ulaw for those carriers. If I make an extension to extension call - there is no audio at all in either direction. All my extensions are set to use G729a (I have tried changing that though to see if it would fix it). I am fairly sure it is not a transcoding issue - as the server transcodes for the inbound/outbound calls. You really need to tell us more! At a pure guess however I'd say you have SIP extensions with canreinvite set to true. Your internal network however does not permit rtp traffic between the handsets. Tim Panton www.mexuar.net www.westhawk.co.uk/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dial plan constructions suggestions?
On Tue, 23 Jan 2007, Ed W wrote: Can I ask for some advice on dial-plan construction please I have setup my dialplan to use 9 to get a zap trunk, leaving everything else for internal extensions. However, this creates a problem in that my callerid is correct, but doesn't work to re-dial the incoming caller. So if I simply click missed calls on my Snom phone and hit redial then it tries to dial an internal extension. So I then setup Asterisk to add a 9 to the incoming callerid for all calls which come via the Zap trunk, but now this creates some issues with applications like Snapanumber and perhaps HudLite, which are trying to map the caller ID to numbers in the addressbook (and I don't really want my internal Outlook address books to have everyone listed with a 9 in front of their number) How are others handling this? There was a thread about this not too long ago, so the archives may have a bit more on it... The way I handle it is by forcing the caller to dial the full number starting with zero (normally 10 or 11 digits in the UK - which I'm guessing you're from too) Zero is the new 9 ;-) This mimics the way mobile phones work here too where you need to dial the full number with the leading zero, even if you'd think it's a local call. Incoming caller-id always provides you with the full 10 or 11 digit number with the zero (or with the country code prefix when that works) The ability to dial the full number in the UK has been avalable for some years now... In my dialplan I look for a leading zero, and when I see one, after premissions checks, etc. I just route the outgoing call with the zero to the relevant zap device. I just tell my clients that they now have to dial the full number including STD code - just like they do on their mobiles. I still provide a traditional '9' for an outside line though - there are a few numbers in the UK that don't start with a leading 0 - 'operator' (100), directory enquriries (118xxx) and a few others for reporting faults, last number dialled and so on. So if they like they can dial 9 then the 5 or 6 digit local number, or just dial 0 for everything. Don't forget to add rules for 999 (and ) to force out-dialling to 999 via a Zap line... And I guess you could eliminate the leading 9 too, if you had explicit entries in the dialplan for these non-zero numbers (or to deny them!) Gordon ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 7 points of comparison Polycom 430/501 and Aastra 480i. Which one to choose ?
On Mon, 2007-01-22 at 16:33 -0600, Jay Moore wrote: The only issue I have with the 480i, is that it's a little unintuitive in how to disable the X missed calls option. There's no option in the web-interface (I'm told one is coming, however), so you have to manually edit a .cfg file and send the info back to the phone. Any chance of sharing the config line, my wife sees missed calls and starts panicking. -- Dave Cotton [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] PRI/Q.sig between Cisco Nortel
Hello, I am using a Cisco-2,811 as a gateway between the Asterisk PBX and our Nortel TX-1 university's PBX. It is working but no names are exchanged. From the debug mode I see that the Cisco sends the display name (which does not appear on the Nortel's phones) and the Nortel does not bother to send it at all. I recall that when I had a pilot with Cisco CCM two years ago we had to set the siganlling to ESGF on the Nortel and use MGCP on Cisco (since MGCP protocol forces the Cisco to use ESGF signalling). We could not use H.323 as it forces the Cisco to use ISGF. I suspect that SIP is the same, but setting ISGF signalling on Nortel doesn't help. Anyone had some luck with this configuration? Thanks, __Yehavi: ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No Mailbox Prompt
David wrote: Is it possible to add a Playback line before the hangup (or before the voicemail) in case someone reaches an extension that doesn't have an active mailbox? Something like: exten = _123105.,2,Playback(no-box,noanswer) You need to use the following command: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+MailboxExists Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] tdm400p not working with brazilian lines
Hi Giorgio, Which telco are you using? I never used a TDM 400 card here in Brazil but I have successfully used X100P and A200 with pretty much default settings. I suggest leaving out caller ID and echo cancellation paremeters for debuging. Regards Bernardo -Original Message- From: Giorgio Incantalupo [EMAIL PROTECTED] Subj: Re: [asterisk-users] tdm400p not working with brazilian lines Date: Tue 23 Jan 2007 8:15 Size: 1K To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Hi Tzafrir, the caller hear the right tone but when the calle picks up the phone Asterisk hangs up. The caller hears the call has answered and hung up in less than a second. We tried the reversepolarity parameter...it initially seemed to work but after some call had the same problem. We think it may depend from telco line but has no info about brazilian telcos...that's why I'm searching for some guy who knows brazilian analog lines. Giorgio Tzafrir Cohen wrote: On Mon, Jan 22, 2007 at 03:16:54PM +0100, Giorgio Incantalupo wrote: Hi, I'm installing an Asterisk box with a TDM2400P in Brazil. I can make analog phones work while lines are not working. What does happend when you try to ring or when a call comes in? Since I do not know anything about brazilian lines, is there anybody who can tell me what is wrong/missing in my conf files (below)? TIA Giorgio _zaptel.conf:_ fxoks=9-16 fxsks=17-24 defaultzone=br loadzone=br* * _zapata.conf:_ context = inbound_zap echocancel = 128 echocancelwhenbridged = yes echotraining = 200 language = br signalling = fxo_ks callerid = Christina 102 channel = 9-16 context = outbound_zap canpark = yes echocancel = 128 echocancelwhenbridged = yes echotraining = 200 faxdetect = both language = br musiconhold = native signalling = fxs_ks callerid = asreceived channel = 17-24 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Can't find asterisk.ctl under CentOS installation
If you are not running asterisk as root, then change the permissions on the /var/run/asterisk folder and below to allow you user access. Also, I seem to remember that on one release, the entry in asterisk.conf did not match safe_asterisk. So the ctl file was NOT where asterisk was looking. -- -- Steven http://www.glimasoutheast.org Devraj Mukherjee [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] Hi Everyone, I recently upgraded to Asterisk 1.4 using the RPMS at ATrpms.net on CentOS 4.4, Asterisk starts up but when I start the console it reports this error and drops out. Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist)? I have checked to see that the file asterisk.ctl actually exists. Any suggestions? -- I never look back darling, it distracts from the now, Edna Mode (The Incredibles) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to exit from console?
Thanks to all. All OK now. I thought that -c option is equivalent to starting an asterisk daemon and connecting to it. Obviously I was wrong. Thanks again, Rudolf On 1/23/07, Doug Lytle [EMAIL PROTECTED] wrote: Rudolf Ladyzhenskii wrote: Hi, all Stupid question, but how do you exit asterisk console without stopping the asterisk? If you start Asterisk without any options: asterisk And then reconnect to it via the -r option asterisk -r Then typing exit on the console will exit without stopping Asterisk. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to exit from console?
On Tue, Jan 23, 2007 at 11:41:23AM +, Marco Mouta wrote: Try safe_asterisk , for an easy way to start asterisk in background, a plain 'asterisk' is even better and safer. asterisk -U asterisk . is better. /etc/init.d/asterisk start is similar. and then connect with asterisk process running asterisk -rx 'asterisk -r' Now you can use exit, And the difference is that 'asterisk -r' just opens a remote terminal to the Asterisk process. Thus you can exit from it. You can't simply exit from the main Asterisk process: you have to shut it down (shut down your PBX). -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DIALSTATUS and HANGUPCAUSE extensions such as s-BUSY
exten = s,2,Goto(s-${DIALSTATUS})
ref:
http://www.voip-info.org/wiki/view/DIALSTATUS
http://www.voip-info.org/wiki-Asterisk+variable+DIALSTATUS
I also use HANGUPCAUSE in some circumstances.
exten = s,2,Goto(s-${HANGUPCAUSE})
ref:
http://www.voip-info.org/wiki/index.php?page=Asterisk+Variable+HANGUPCAUSE
--
--
Steven
http://www.glimasoutheast.org
Barzilai Spinak [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED]
I've seen several examples that use extensions such as;
s-BUSY
s-NOANSWER
etc...
It's more or less evident what they do, but I've searched for some FORMAL
documentation everywhere and have found nothing.
Do they work for anything else than s-? (I think I've seen other examples,
but can't find them now)
Are they standard in any way?
What are the allowed values after the dash?
In which version were they introduced?
etc...
(please no replies explaining me how s-BUSY matches when the start
extension is set busy or trivial explanations like that)
BarZ
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[asterisk-users] Re: Can't find asterisk.ctl under CentOS installation
On Tue, Jan 23, 2007 at 09:27:03PM +1100, Devraj Mukherjee wrote: I recently upgraded to Asterisk 1.4 using the RPMS at ATrpms.net on CentOS 4.4, Asterisk starts up but when I start the console it reports this error and drops out. Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist)? I have checked to see that the file asterisk.ctl actually exists. Any suggestions? Have you tried asterisk -r as root or as a non-root user? See below for details. On Tue, Jan 23, 2007 at 08:10:29AM -0500, Steven wrote: If you are not running asterisk as root, then change the permissions on the /var/run/asterisk folder and below to allow you user access. asterisk on ATrpms runs as asterisk:asterisk and the folder belongs to that uid/gid: # ls -ld /var/run/asterisk drwxr-xr-x 2 asterisk asterisk 4096 Jan 16 17:30 /var/run/asterisk Also, I seem to remember that on one release, the entry in asterisk.conf did not match safe_asterisk. So the ctl file was NOT where asterisk was looking. asterisk.conf does point to the same folder. # grep astrundir /etc/asterisk/asterisk.conf astrundir = /var/run/asterisk And safe_asterisk/asterisk do use that file according to lsof asterisk3u unix 0x8800332dfc00 213094 /var/run/asterisk/asterisk.ctl If I call asterisk -r as root it succeeds, if as another user it will give Devraj's error message. That's probably how it is supposed to work, or not? -- Axel.Thimm at ATrpms.net pgpS41UvgFA8z.pgp Description: PGP signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Operate on registrations
23 jan 2007 kl. 10.42 skrev yusuf: Hi, I have a bunch of SIP phones(behind NAT) registering on my * box. I want to find out when they register and de-register. I also want to operate on it, so when they register/de-register, I want to insert calldate into a mysql DB, etc. Maybe this will help me when, for instance a user tries to register but has the wrong username/password. Now I am aware of regcontext, but it only creates a 1,NoOP for that user, I want it to execute that, so I can have this maybe: exten = 666,2,AGI(Registraion.agi) so when my users register 666,1,NoOp will be created and execution can start there. Any Ideas on how I can get something like this? We do send events for registration/deregistration to the manager interface (AMI). I think that's the best way to handle this. /O ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] problems with dtmf
Hello: I have a problem when a person call a queue and the agent answer the call. If the caller press any key during the dialog the agent hear a continue sound like dtmf. Any ideas? Thanks. Patricio. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Can't find asterisk.ctl under CentOS installation
On Tue, Jan 23, 2007 at 02:48:07PM +0100, Axel Thimm wrote: If I call asterisk -r as root it succeeds, if as another user it will give Devraj's error message. That's probably how it is supposed to work, or not? Just a thought: shouldn't the asterisk user be allowed write access to that control socket? Or maybe the asterisk group? (for quickdirty shell scripts) -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Dial plan constructions suggestions?
On Tue, 2007-01-23 at 06:11 -0700, Ed W [EMAIL PROTECTED] wrote: Can I ask for some advice on dial-plan construction please I have setup my dialplan to use 9 to get a zap trunk, leaving everything else for internal extensions. However, this creates a problem in that my callerid is correct, but doesn't work to re-dial the incoming caller. So if I simply click missed calls on my Snom phone and hit redial then it tries to dial an internal extension. So I then setup Asterisk to add a 9 to the incoming callerid for all calls which come via the Zap trunk, but now this creates some issues with applications like Snapanumber and perhaps HudLite, which are trying to map the caller ID to numbers in the addressbook (and I don't really want my internal Outlook address books to have everyone listed with a 9 in front of their number) How are others handling this? I have considered simply dropping the prefix digit and working around any clashes in internal and external numbers (not very hard). I had the same situation, in that I wanted to be able to use the Voicemail 'dial back' feature, and had a few phones with internal CID-based dial features, that I wanted to be allowed to be used. Your normal context is set up to operate with a '9' (or whatever) in front; so it is clear that you will need a different context from which to dial, a context that doesn't have the '9' at the beginning. For voicemail, the solution is simple; you create a context much like your normal outgoing context, except, without the '9' in front. Then, in voicemail.conf, you make use of: ; dialout=fromvm; Context to dial out from [option 4 ; from the advanced menu] ; if not listed, dialing out will not be permitted ; callback=fromvm ; Context to call back from ; if not listed, calling the sender back will not ; be permitted by declaring your mailbox as so: 80 = ,Alex Murphy,,,callback=fromvmhome|dialout=fromvmhome| saycid=yes|tz=mountain|review=yes (stupid mail package folds lines!) --- For those phones with a CID list of their own, and autodialers, you merely associate them with a context that allows that... in zapata.conf, you might set up: context = fromSeanUniden callerid=UnidenPowerMax2.4Ghz5 channel = 5 Or whatever's appropriate for your setup. The main idea is to set the 'context' so something special for that phone. --- hope this helps...! murf smime.p7s Description: S/MIME cryptographic signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] bad gateway error on snom display
Hi, sometimes I get a bad gateway error on my snom phone model 320 and 360. Has anyone else got the same message? How to solve it? TIA Giorgio Incantalupo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Streaming audio file while working in background ?
El lun, ene 22 de 2007 a las 16:58 -0700, Darren Nay comentaba: Ideally I would like to be able to play an audio file to the caller while making outbound calls in the background (via the Dial app) and then discontinue the audio file stream and bridge the calls once an outbound call is connected. Maybe you can execute a macro when the called party answers and still provide music on hold or something to the caller, this is done with the opcion 'M' of the Dial app. http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial Look at the Example Nr 2. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] stress-test realtime voicemail with sipp
We are in the process of implementing realtime voicemail. I was wanting to stress-test the system to see if or when it would fall over. Is it possible to use sipp to create say 250 calls, each of which leaves a message in the voicemail ? My dialplan is currently [default] exten = stress,1,Answer() exten = stress,2(vm),Voicemail(|su) exten = stress,3,Hangup() however, if I use sipp to test this, I get [Jan 23 14:43:51] WARNING[22782]: app.c:599 __ast_play_and_record: No audio available on SIP/sipp-b7c274b0?? I suspect that's because sipp itself is not sending audio. Is there any tricks I can do in the dialplan to get an extension to answer sipp and then send it to voicemail, but play some audio for the voicemail ? Thanks. Julian. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.4 Polycom buddy status
I'm running into an issue w/ Buddy status on Polycom IP650 phones using buddy status (with SIP Hints) on Asterisk 1.4. Sometimes the status on the phones will stick in the busy status. I have noticed that I can call that extension the status will reset (sometimes). Anyone else encountered this or anything similar. -Chris ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] stress-test realtime voicemail with sipp
El mar, ene 23 de 2007 a las 14:44 +, Julian Lyndon-Smith comentaba: however, if I use sipp to test this, I get [Jan 23 14:43:51] WARNING[22782]: app.c:599 __ast_play_and_record: No audio available on SIP/sipp-b7c274b0?? I suspect that's because sipp itself is not sending audio. Why don't you use sipp with pcap support enabled? http://sipp.sourceforge.net/doc/reference.html You can modify a little bit some of the integrated scenarios to allow sipp to interoperate with your voicemail extension. http://sipp.sourceforge.net/doc/reference.html#UAC+with+media ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to exit from console?
Rudolf Ladyzhenskii wrote: Hi, all Stupid question, but how do you exit asterisk console without stopping the asterisk? Tried quit and exit: *CLI exit No such command 'exit' (type 'help' for help) *CLI quit No such command 'quit' (type 'help' for help) *CLI Any other ideas? I started asterisk with -cg option. Same problem if use asterisk -r to connect. Can not exit. Any ideas? Thanks, Rudolf ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ctrl-c :-) signature.asc Description: OpenPGP digital signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: DIALSTATUS and HANGUPCAUSE extensions such ass-BUSY
Also note: it is not just s-
If your extension is 5111, you can use
exten = 5111,2,Goto(5111-${HANGUPCAUSE})
--
--
Steven
http://www.glimasoutheast.org
Steven [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED]
exten = s,2,Goto(s-${DIALSTATUS})
ref:
http://www.voip-info.org/wiki/view/DIALSTATUS
http://www.voip-info.org/wiki-Asterisk+variable+DIALSTATUS
I also use HANGUPCAUSE in some circumstances.
exten = s,2,Goto(s-${HANGUPCAUSE})
ref:
http://www.voip-info.org/wiki/index.php?page=Asterisk+Variable+HANGUPCAUSE
--
--
Steven
http://www.glimasoutheast.org
Barzilai Spinak [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED]
I've seen several examples that use extensions such as;
s-BUSY
s-NOANSWER
etc...
It's more or less evident what they do, but I've searched for some FORMAL
documentation everywhere and have found nothing.
Do they work for anything else than s-? (I think I've seen other examples,
but can't find them now)
Are they standard in any way?
What are the allowed values after the dash?
In which version were they introduced?
etc...
(please no replies explaining me how s-BUSY matches when the start
extension is set busy or trivial explanations like that)
BarZ
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Re: [asterisk-users] stress-test realtime voicemail with sipp (Solved)
Thanks Victor for the heads up. I've got it to work with the following: [default] exten = stress,1,Answer() exten = stress,2(vm),Voicemail() exten = stress,3,Hangup() and a sipp command line of ./sipp -d 4 -r 5 -t un -sn uac_pcap -l 50 -m 250 -s stress 127.0.0.1 this created 250 voicemail messages (with 50 simultaneous calls) leaving a 6-7 second voicemail (using .wav, .WAV and .gsm) I'm *really* going to try and hurt it now ;) Julian Victor Toofic wrote: El mar, ene 23 de 2007 a las 14:44 +, Julian Lyndon-Smith comentaba: however, if I use sipp to test this, I get [Jan 23 14:43:51] WARNING[22782]: app.c:599 __ast_play_and_record: No audio available on SIP/sipp-b7c274b0?? I suspect that's because sipp itself is not sending audio. Why don't you use sipp with pcap support enabled? http://sipp.sourceforge.net/doc/reference.html You can modify a little bit some of the integrated scenarios to allow sipp to interoperate with your voicemail extension. http://sipp.sourceforge.net/doc/reference.html#UAC+with+media ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 7 points of comparison Polycom 430/501 and Aastra480i. Which one to choose ?
I am going to weigh in on this one because is an issue near and dear to me ... In my fleet, I have mostly Cisco 7960 phones ... We have a few of the Polycom 501's and a few of the Aastra 480i phones ... My personal preference is the Aastra but not based on your priorities ... Addressing your requirements in order of preference ... Sound quality ... 1) Polycom/Cisco (toss up, they are roughly equal) 2) Aastra (its close, but no cigar) Complete product line 1) Cisco (about every model you could ask for) 2) Polycom (missing the wireless model and have not seen a receptionist model) 3) Aastra (good quality basic and executive phones only - wireless is married to an executive phone and is NOT WiFi) POE 1) Cisco, Polycom, Aastra (toss up, they are roughly equal) Backlit LCD 1) Aastra (though its so dim its useless) 2) Polycom, Cisco (not available unless you go the new Cisco models) Speaker phone 1) Polycom, Cisco (toss up, they are equal) 2) Aastra (close in quality but no cigar) Dual LAN NIC 1) Cisco, Polycom, Aastra (toss up, they all have it) As to my own requirements ... I place a very high priority on availability of driver updates and support ... Go to the Cisco and Polycom web sites and TRY to download their latest SIP firmware ... Once you are totally frustrated, go to the Aastra site and do the same ... You will find that user manuals and software updates are readily available for Aastra phones without having to jump through ridiculous hoops ... CASE CLOSED ... From now on I am buying Aastra ... The Aastra phone is very nice ... I have figured out how to make ALL the buttons work, it looks impressive and has much better sound quality than the cheapies ... It is NOT as high quality in construction or sound as the Cisco or Polycom but it is VERY close ... But with their superior support offering, Aastra tips the scales ... You will not go wrong if you choose any of the three I mention ... I have been doing this now for about five years and have tried just about every new phone model to hit the market at one point or another ... I dearly love the Cisco phones and if it was not for their unfortunate support policies, that is the only phone I would use ... When I went looking for a replacement, I tried Polycom, liked them but was once again confronted with high handed support policies ... So even though they were not my first choice, the boys at Aastra have earned my business ... G.Hendershot -Original Message- From: Vikas [mailto:[EMAIL PROTECTED] Sent: Monday, January 22, 2007 5:12 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] 7 points of comparison Polycom 430/501 and Aastra480i. Which one to choose ? I need to provide a 80 people office with VOIP. I want to commit to one vendor Polycom or Aastra. Price of the phones is not a factor in the decision. The quality of the phones is the factor. Some of the features that I am evaluating on are: (arranged in order of priority) 1. Sound quality 2. complete product line with conference phone and receptionist phone (not on Aastra) 3. cordless (not on 501/430) 4. backlit LCD (not on 501/430) 5. Inbuilt POE (not on 501) 6. speaker phone 7. 2 network ports. Which one will you choose ? Vikas ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 Polycom buddy status
23 jan 2007 kl. 16.09 skrev Chris Bullock: I'm running into an issue w/ Buddy status on Polycom IP650 phones using buddy status (with SIP Hints) on Asterisk 1.4. Sometimes the status on the phones will stick in the busy status. I have noticed that I can call that extension the status will reset (sometimes). Anyone else encountered this or anything similar. I've seen reports on it, but haven't been able to repeat this. I need to know a way to force this to happen, repeatably. If I can get that, I can propably trace it and fix it. It can also happen if you have packet loss in the network, of course. /O ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to exit from console?
I personally run asterisk in a screen session. Gets rid of this problem and makes things a lot easier. - Original Message - From: Marco Mouta To: [EMAIL PROTECTED] ; Asterisk Users Mailing List - Non-Commercial Discussion Sent: Tuesday, January 23, 2007 1:41 PM Subject: Re: [asterisk-users] How to exit from console? Try safe_asterisk , for an easy way to start asterisk in background, and then connect with asterisk process running asterisk -rx Now you can use exit, and by the way you may look on wiki diferent ways to run asterisk. On 1/23/07, Rudolf Ladyzhenskii [EMAIL PROTECTED] wrote: Hi, all Stupid question, but how do you exit asterisk console without stopping the asterisk? Tried quit and exit: *CLI exit No such command 'exit' (type 'help' for help) *CLI quit No such command 'quit' (type 'help' for help) *CLI Any other ideas? I started asterisk with -cg option. Same problem if use asterisk -r to connect. Can not exit. Any ideas? Thanks, Rudolf ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] stress-test realtime voicemail with sipp
As far as I know: You need to compile sipp with media streaming and authentication or if you just want first to test you may provide an extension named service in the context defined in general section of your sip conf for external calls coming to your asterisk server without authentication: http://sipp.sourceforge.net/doc/reference.html#Installing+SIPp - *With PCAP playhttp://sipp.sourceforge.net/doc/reference.html#pcapplayand without authenticationhttp://sipp.sourceforge.net/doc/reference.html#authenticationsupport *: # gunzip sipp-xxx.tar.gz # tar -xvf sipp-xxx.tar # cd sipp # make pcapplay - *With PCAP playhttp://sipp.sourceforge.net/doc/reference.html#pcapplayand authenticationhttp://sipp.sourceforge.net/doc/reference.html#authenticationsupport *: # gunzip sipp-xxx.tar.gz # tar -xvf sipp-xxx.tar # cd sipp # make pcapplay_ossl Example: - Sipp being used as a SIP user agent Client: - Call Duration 1ms - Dialing Calls with RTP using ulaw ./sipp -sf uac_pcap.xml -d 1 192.168.34.6 -trace_err Where this IP is my * . Hope this helps, Plse provid some feedback. I would like also to learn from community how to understand Load average results with Top command while incrementing calls dial from sipp to asterisk, and how to determine max calls on Asterisk. This max calls is defined when Sipp calls to * starts being discarded? Best regards, Marco Mouta On 1/23/07, Julian Lyndon-Smith [EMAIL PROTECTED] wrote: We are in the process of implementing realtime voicemail. I was wanting to stress-test the system to see if or when it would fall over. Is it possible to use sipp to create say 250 calls, each of which leaves a message in the voicemail ? My dialplan is currently [default] exten = stress,1,Answer() exten = stress,2(vm),Voicemail(|su) exten = stress,3,Hangup() however, if I use sipp to test this, I get [Jan 23 14:43:51] WARNING[22782]: app.c:599 __ast_play_and_record: No audio available on SIP/sipp-b7c274b0?? I suspect that's because sipp itself is not sending audio. Is there any tricks I can do in the dialplan to get an extension to answer sipp and then send it to voicemail, but play some audio for the voicemail ? Thanks. Julian. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] weird undocumented extensions such as s-BUSY
Barzilai Spinak wrote: I've seen several examples that use extensions such as; s-BUSY s-NOANSWER etc... It's more or less evident what they do, but I've searched for some FORMAL documentation everywhere and have found nothing. Do they work for anything else than s-? (I think I've seen other examples, but can't find them now) Are they standard in any way? What are the allowed values after the dash? In which version were they introduced? etc... (please no replies explaining me how s-BUSY matches when the start extension is set busy or trivial explanations like that) Try looking in extensions.conf.sample ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dial plan constructions suggestions?
Hi There was a thread about this not too long ago, so the archives may have a bit more on it... The way I handle it is by forcing the caller to dial the full number starting with zero (normally 10 or 11 digits in the UK - which I'm guessing you're from too) Yes, I use something similar on another box, but there I support shorter dial codes as well. It's not to hard to make 8 dial 0208 or 7 dial 0207, etc. I happen to also map some of the 1xx codes across as well. It's still not a complete solution though because on this other box I have a business line and a personal line and I send calls to different lines based on the type of call (or more usually the time of day...). I want to have seperate billing basically. When the call comes in it makes sense to have the caller tagged with (in my dialplan) 9 for a personal call, and I use 3 (for no good reason) for my business line. I actually have one phone which defaults to business line if I don't add a prefix, another DECT phone which is my personal phone, but I can see on either where the call is coming from and also force the call to use a different route just by dialing the prefix. Basically it's tricky. I do already use custom ring tones for each line, so I guess I could drop the prefix, but it's nice to have it so that I can see at a glance whether it's a business call or not... Any other suggestions? Any suggestions on other software than Snap which does callerId lookup from Thunderbird (not Outlook). For example is HUDLite ever going to support Thunderbird...? Cheers Ed W ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Dial plan constructions suggestions?
Hi I had the same situation, in that I wanted to be able to use the Voicemail 'dial back' feature, and had a few phones with internal CID-based dial features, that I wanted to be allowed to be used. Your normal context is set up to operate with a '9' (or whatever) in front; so it is clear that you will need a different context from which to dial, a context that doesn't have the '9' at the beginning. I appreciate your point, but it's not that hard to avoid having the 9 prefix at all (in a simple dialplan at least). So to be honest one might as well dump the whole dial 9 thing completely in the scenario you describe? I think the solution here is really that the CID type applications become aware of prefix digits and strip them. Anyone know of good solutions to this? Any backend solutions to get Asterisk to hook into Exchange server etc? Cheers Ed W ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Rhino cards lock up system -- anyone else ever seen this?
Hi Folks, Struggling with a new * installation with 2 Rhino R2T1 cards. For some reason, the system is locking up tight when you run ztcfg to configure the card(s). Configuration is asterisk 1.2.14, zaptel 1.2.12, and rhino's 1.05rxt1 drivers. The cards seem to load fine with a modprobe rxt1, but once you run ztcfg -vvv, the system will lock up within a few seconds, no errors reported in logs or console. I'm stumped, Rhino is stumped, and I haven't seen any other threads of this nature. -- Barry D. Hassler ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] How to exit from console?
you have to start it with no options in order to -r into and quit out of it _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dovid B Sent: Tuesday, January 23, 2007 10:46 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How to exit from console? I personally run asterisk in a screen session. Gets rid of this problem and makes things a lot easier. - Original Message - From: Marco mailto:[EMAIL PROTECTED] Mouta To: [EMAIL PROTECTED] ; Asterisk Users Mailing List - mailto:asterisk-users@lists.digium.com Non-Commercial Discussion Sent: Tuesday, January 23, 2007 1:41 PM Subject: Re: [asterisk-users] How to exit from console? Try safe_asterisk , for an easy way to start asterisk in background, and then connect with asterisk process running asterisk -rx Now you can use exit, and by the way you may look on wiki diferent ways to run asterisk. On 1/23/07, Rudolf Ladyzhenskii [EMAIL PROTECTED] wrote: Hi, all Stupid question, but how do you exit asterisk console without stopping the asterisk? Tried quit and exit: *CLI exit No such command 'exit' (type 'help' for help) *CLI quit No such command 'quit' (type 'help' for help) *CLI Any other ideas? I started asterisk with -cg option. Same problem if use asterisk -r to connect. Can not exit. Any ideas? Thanks, Rudolf ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AEL parse failure on 1.2.14
Am I doing something really stupid in this AEL macro, or is nesting an
'if' inside a 'switch', inside an 'if' not supported in the 1.2 AEL
parser?
macro stdexten( ext , dev ) {
// First determine if the SIP peer is registered here
Set(aretheyhere=${SIPPEER(${ext}:status)});
if(${aretheyhere:0:2}) == OK) {
MixMonitor(${UNIQUEID}.wav|b);
Dial(${dev},30);
switch(${DIALSTATUS}) {
case BUSY:
MailboxExists(${ext});
if(${VMBOXEXISTSSTATUS} != SUCCESS) {
Busy(5);
};
Voicemail(b${ext});
Hangup();
break;
default:
MailboxExists(${ext});
if(${VMBOXEXISTSSTATUS} != SUCCESS) {
Congestion(5);
};
Voicemail(u${ext});
Hangup();
break;
};
};
};
When I do an AEL reload, I get
2007-01-23 16:11:31 WARNING[10795]: pbx_ael.c:102 __grab_token: Syntax error at
line 370 of 'extensions.ael', too many closing braces!
-- Registered extension context 'macro-stdexten'
-- Added extension 's' priority 1 to macro-stdexten
-- Added extension 's' priority 2 to macro-stdexten
-- Added extension 's' priority 3 to macro-stdexten
-- Added extension 's' priority 4 to macro-stdexten
2007-01-23 16:11:31 WARNING[10795]: pbx_ael.c:102 __grab_token: Syntax error at
line 371 of 'extensions.ael', too many closing braces!
2007-01-23 16:11:31 NOTICE[10795]: pbx_ael.c:1146 handle_root_token: Unknown
root token '}'
Cheers,
Gavin.
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[asterisk-users] IGNORE: AEL parse failure on 1.2.14
Doh!
macro stdexten( ext , dev ) {
// First determine if the SIP peer is registered here
Set(aretheyhere=${SIPPEER(${ext}:status)});
if(${aretheyhere:0:2}) == OK) {
^^^ errant close-bracket
Sorry for the noise (twice).
gdh
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Re: [asterisk-users] stress-test realtime voicemail with sipp
23 jan 2007 kl. 16.07 skrev Victor Toofic: El mar, ene 23 de 2007 a las 14:44 +, Julian Lyndon-Smith comentaba: however, if I use sipp to test this, I get [Jan 23 14:43:51] WARNING[22782]: app.c:599 __ast_play_and_record: No audio available on SIP/sipp-b7c274b0?? I suspect that's because sipp itself is not sending audio. Why don't you use sipp with pcap support enabled? http://sipp.sourceforge.net/doc/reference.html You can modify a little bit some of the integrated scenarios to allow sipp to interoperate with your voicemail extension. http://sipp.sourceforge.net/doc/reference.html#UAC+with+media Easier is to use another ASterisk server, or two... /O ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to exit from console?
Rudolf Ladyzhenskii wrote: Hi, all Stupid question, but how do you exit asterisk console without stopping the asterisk? Tried quit and exit: *CLI exit No such command 'exit' (type 'help' for help) *CLI quit No such command 'quit' (type 'help' for help) *CLI Any other ideas? I started asterisk with -cg option. Same problem if use asterisk -r to connect. Can not exit. Many unix programs you can exit with ctrl d - asterisk console does also. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Can't find asterisk.ctl under CentOS installation
On Tue, Jan 23, 2007 at 03:54:49PM +0200, Tzafrir Cohen wrote: On Tue, Jan 23, 2007 at 02:48:07PM +0100, Axel Thimm wrote: If I call asterisk -r as root it succeeds, if as another user it will give Devraj's error message. That's probably how it is supposed to work, or not? Just a thought: shouldn't the asterisk user be allowed write access to that control socket? Or maybe the asterisk group? The asterisk user is allowed, too, of course, the group not (yet). (for quickdirty shell scripts) I think that makes very much sense. The socket is created by asterisk, is there a parameter to specify permissions/umask of that socket? -- Axel.Thimm at ATrpms.net pgpayWA7IHV3n.pgp Description: PGP signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Can't find asterisk.ctl under CentOS installation
On Tue, Jan 23, 2007 at 06:20:08PM +0100, Axel Thimm wrote: On Tue, Jan 23, 2007 at 03:54:49PM +0200, Tzafrir Cohen wrote: On Tue, Jan 23, 2007 at 02:48:07PM +0100, Axel Thimm wrote: If I call asterisk -r as root it succeeds, if as another user it will give Devraj's error message. That's probably how it is supposed to work, or not? Just a thought: shouldn't the asterisk user be allowed write access to that control socket? Or maybe the asterisk group? The asterisk user is allowed, too, of course, the group not (yet). (for quickdirty shell scripts) I think that makes very much sense. The socket is created by asterisk, is there a parameter to specify permissions/umask of that socket? Looks like all there is needed is to uncomment the following line in the default config file: [files] astctlpermissions = 0660 But since upstream defaults to not do so and only have this done by the user, I wouldn't like to change this policy on the package level. -- Axel.Thimm at ATrpms.net pgpOJ2MiE2aSI.pgp Description: PGP signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue and Interface time out
Okay, that makes sense. I wasn't thinking about the SIP driver needing to be told to track the peer's status. I assumed it just did that. So now there's a new problem. The Queue application doesn't always clear the member interface's status after completing a call. The SIP peer no longer has an active channel but the queue will still show the member 'In use'. The occurrence of this is erratic and I have been unable to determine any commonalities among the callers or members other than that it happens to all members. Connecting to the peer outside of the queue will clear the status. Any ideas? Thanks, James Watkins, Bradley wrote: What it actually does is tell the SIP channel driver to track whether or not any given peer has a call to it. It can then subsequently inform the Queue application so that another call will not be given to that user. If you did not have the ringinuse=no in your queue definition, you would then be able to receive up to 5 simultaneous calls (after five, then the SIP channel driver would return busy and Queue wouldn't be able to dial that peer). Regards, - Brad From: [EMAIL PROTECTED] on behalf of James Fromm Sent: Fri 1/19/2007 6:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Queue and Interface time out That worked. I don't understand what call-limit has to do with this. I set it to 5. Why does that keep the member interface from getting a second call from the Queue application? I would think it would allow the member interface to get up to 5 calls. The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re:sip giving problems, please help.
SIP/15552830438-990b doesn't seem to be a valid channel name, try doing an fsck. On 9/4/06, Ma Zhiyong [EMAIL PROTECTED] wrote: Yes, I also get these problems occasionally Sep 4 17:44:49 WARNING[1365]: channel.c:787 channel_find_locked: Avoided deadlock for '0x8224468', 10 retries! Sep 4 17:44:49 WARNING[1364]: channel.c:787 channel_find_locked: Avoided deadlock for '0x8224468', 10 retries! Sep 4 17:52:15 WARNING[1597]: ast_expr2.fl:183 ast_yyerror: ast_yyerror(): syntax error: syntax error, unexpected TOK_LT, expecting TOK_MINUS or TOK_COMPL or TOK_LP or TOKEN; Input: 60 ^ Sep 4 17:52:15 WARNING[1597]: ast_expr2.fl:187 ast_yyerror: If you have questions, please refer to doc/README.variables in the asterisk source. Sep 4 17:52:15 WARNING[1597]: ast_expr2.fl:183 ast_yyerror: ast_yyerror(): syntax error: syntax error, unexpected TOK_LT, expecting TOK_MINUS or TOK_COMPL or TOK_LP or TOKEN; Input: 120 ^ Sep 4 17:52:15 WARNING[1597]: ast_expr2.fl:187 ast_yyerror: If you have questions, please refer to doc/README.variables in the asterisk source. Sep 4 18:50:49 ERROR[1290]: chan_sip.c:11346 sipsock_read: We could NOT get the channel lock for SIP/gw-442744f0! Sep 4 18:50:49 ERROR[1290]: chan_sip.c:11347 sipsock_read: SIP MESSAGE JUST IGNORED: BYE Sep 4 18:50:49 ERROR[1290]: chan_sip.c:11348 sipsock_read: BAD! BAD! BAD! Sep 4 18:50:51 ERROR[1290]: chan_sip.c:11346 sipsock_read: We could NOT get the channel lock for SIP/gw-442744f0! Sep 4 18:50:51 ERROR[1290]: chan_sip.c:11347 sipsock_read: SIP MESSAGE JUST IGNORED: BYE Sep 4 18:50:51 ERROR[1290]: chan_sip.c:11348 sipsock_read: BAD! BAD! BAD! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Facundo Ameal. famealatgmaildotcom Linux User #395088 Share your knowledge, use free software. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] weird undocumented extensions such as s-BUSY
So, in sum. It's just an Asterisk idiom or best(?) practice that has become somewhat common. Every time I need to do the smallest thing in Asterisk I have to Google for 3 hours and read voip-info for the more hours, and then read old mailing list post for 3 more, until I can filter out *real information* from the background noise. Maybe it's just me... I don't settle with the first solution that SeemsToWorkForNow, even though I have no idea Why or How Ah... the wonders of Early 21st Century Web Fashion and documenting everything in big lumps of Little-Ultra-Hyper-Linked-Wiki-pages with contradictory and obsolete info Hopefully it will end some day and the world will come back to its senses. BarZ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] * 1.0.9 Voicemail record name does not playback in Directory()
On * 1.0.9 User logs into voicemail, dials Option Zero, then Option Three. Records name, accepts the recording. greet.wav is generated in the user's mailbox. It plays back fine. The Directory app still spells out his name! What am I missing? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] weird undocumented extensions such as s-BUSY
On Tuesday 23 January 2007 1:16 pm, Barzilai Spinak wrote: So, in sum. It's just an Asterisk idiom or best(?) practice that has become somewhat common. Every time I need to do the smallest thing in Asterisk I have to Google for 3 hours and read voip-info for the more hours, and then read old mailing list post for 3 more, until I can filter out *real information* from the background noise. Maybe it's just me... I don't settle with the first solution that SeemsToWorkForNow, even though I have no idea Why or How Ah... the wonders of Early 21st Century Web Fashion and documenting everything in big lumps of Little-Ultra-Hyper-Linked-Wiki-pages with contradictory and obsolete info Hopefully it will end some day and the world will come back to its senses. After you're done hyperventilating, feel free to contribute documentation which you find is meaningful, current and insightful. Open-source in general is very much a get your hands dirty kind of software experience. This means that you are expected to play around, experiment, and ask good questions, ALL without throwing a little tantrum as you just did. If you want current manuals, completely stable software and someone to yell at when your system breaks, Digium offers that, too. It's called Asterisk Business Edition. Otherwise, dig in, experiment and try to leave the place a little cleaner than you left it.. -A. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] weird undocumented extensions such as s-BUSY
Barzilai Spinak wrote: So, in sum. It's just an Asterisk idiom or best(?) practice that has become somewhat common. Every time I need to do the smallest thing in Asterisk I have to Google for 3 hours and read voip-info for the more hours, and then read old mailing list post for 3 more, until I can filter out *real information* from the background noise. Maybe it's just me... I don't settle with the first solution that SeemsToWorkForNow, even though I have no idea Why or How Ah... the wonders of Early 21st Century Web Fashion and documenting everything in big lumps of Little-Ultra-Hyper-Linked-Wiki-pages with contradictory and obsolete info Hopefully it will end some day and the world will come back to its senses. BarZ Pretty much - Not restricted to Asterisk though - rampant in Open Source in general, and to a lesser extent in the computer field as a whole. I have been struggling for a while with no good answer to come to the conclusion that Asterisk 1.4 can't be installed on CentOs 3.8 There is no information associated with the download, such as system requirements That would make it too easy After googling and searching for hours/days, solving one problem compiling Zaptel, breezing through the Libpri, run into another brick wall because some little POS called ptlib-config doesn't exist, and no clue where it is, or SHOULD it even exist, So on to CentOS 4.4. and we'll see what that brings Don't expect the world to come (back?) to its senses either This is as good as it gets! John Novack ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] [OT] Mark Spencer Presents AsteriskNOW on Youtube
Mark Spencer, the original author and founder of Digium, presents AsteriskNOW -- Damián D. Fossi Salas ¡Software Libre hasta el 2 mil siempre! Uso: Debian Etch Kernel 2.6.18-3-686 Ubuntu Edgy Eft Kernel 2.6.15-27-amd64 Ulanix 0.4-14 Kernel 2.6.18-486 FreeBSD 6.2-RC1 Linux User: 188464 GPG Key Fingerprint = EC09 9ABA DFD8 83F0 36F3 CA89 356E 27FD E666 E6A4 Jabber ID: damianfossi en jabberes.org www.damianfossi.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] TDM2400 Hardware Echo Cancel
I have been having the same problems since installing a TDM2400 with hardware echo canceller. The best way to describe the sound is a background crackle or hiss that just can't be filtered out. Increasing the RX gain just makes the problem worse. SIP to SIP calls are flawless. An acquaintance told me the analog line level is too low, but when plugging a regular phone into the line, the signal is plenty loud enough. I am curious if anyone else had similar issues with the TDM2400 card and if they have resolved it. -- Andrew -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Adam Sharples Sent: Tuesday, January 16, 2007 09:00 To: asterisk-users@lists.digium.com Subject: [asterisk-users] TDM2400 Hardware Echo Cancel Good Day List, I'm having some issues with echo cancel on my Asterisk box, and have done extensive reading and have gained some useful pointers from this list but have a couple of hopefully fairly simple questions. The Asterisk box is connected via 20 FXO ports on a TDM2400 with the Hardware echo cancel module. Echo cancel almost works, but the users hear what they describe as a 'crackle' coming back when they talk. I want to tune to echo canceller, but am unsure if any of the options provided have any effect on the hardware module. Do the settings such as echocancel and echotraining in Zapata.conf affect the hardware module? Would I be better removing the hardware module and tuning the software echo canceller? The asterisk box is currently running 1.2.13, with zaptel 1.2. Would you advise upgrading to the newer Zaptel drivers? I don't want to upgrade Asterisk itself just yet. Any help or pointers to documentation regarding the hardware echo cancel module would be greatly appreciated, Thanks, Adam Sharples ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to exit from console?
On Tue, Jan 23, 2007 at 10:55:12AM -0600, Tim Litwiller wrote: Rudolf Ladyzhenskii wrote: Hi, all Stupid question, but how do you exit asterisk console without stopping the asterisk? Tried quit and exit: *CLI exit No such command 'exit' (type 'help' for help) *CLI quit No such command 'quit' (type 'help' for help) *CLI Any other ideas? I started asterisk with -cg option. Same problem if use asterisk -r to connect. Can not exit. Many unix programs you can exit with ctrl d - asterisk console does also. The main asterisk process, when not daemonized, behaves that way, I guess. However asterisk -r does not respect ctrl-d as a hint for end of file (end of standard input) . BTW: there is another way in which asterisk -c is not a standard process: start a xterm/putty window, connect as root and run a new asterisk -c in another terminal run asterisk -rv . Now, what happens there when you close that xterm window without stopping asterisk first: will asterisk exit like any proper interactive program does when it loses its controlling terminal? no. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] OT: High Quality Wireless Headset for Cisco IP Phones and *
Has anyone found a high quality wireless headset that works well with Cisco 7960 IP phones on an asterisk system? I tried the vxxi offering but the sound quality was pretty bad. Since these are pricey, I don't want to sample blindly. Experience appreciated. Thanks, Tom ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] OT: High Quality Wireless Headset for Cisco IPPhones and *
Tom - here are a few suggestions Plantronics 510SL Bluetooth - they make a bundle that comes with a Bluetooth transceiver and a handset lifter. This is nice if you have users that already have Bluetooth headsets for their mobile phones, as it allows you to sync with their office phone as well. http://www.plantronics.com/north_america/en_US/products/cat29880043/cat2 9880054/prod5460016 Plantronics CS70 - Uses DECT for the wireless component http://www.plantronics.com/north_america/en_US/products/cat29880043/cat2 9880054/prod5510016 GN Netcom - GN 9350 - also uses DECT and supports wideband audio http://www.gnnetcom.com/US/EN/MainMenu/Products/Wireless+Solutions/GN+93 50.htm Cory Andrews -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tom Sent: Tuesday, January 23, 2007 3:02 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] OT: High Quality Wireless Headset for Cisco IPPhones and * Has anyone found a high quality wireless headset that works well with Cisco 7960 IP phones on an asterisk system? I tried the vxxi offering but the sound quality was pretty bad. Since these are pricey, I don't want to sample blindly. Experience appreciated. Thanks, Tom ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Dial plan constructions suggestions?
On 1/23/07, Ed W [EMAIL PROTECTED] wrote: I appreciate your point, but it's not that hard to avoid having the 9 prefix at all (in a simple dialplan at least). So to be honest one might as well dump the whole dial 9 thing completely in the scenario you describe? I originally setup without the 9. But, here in the US, we have 711 which interferes with PARK. We also have 511 in some areas. I could figure out a way around everything but the 711. I guess you could probably get around the PARK situation by starting your parking spots at 720 or something. I just went back to the 9. I use a custom caller ID lookup database that I was able to just add strip the 9 in the program itself. As far as callback, all my numbers are prefixed with a company code to indicate which company the caller is calling, so my dialplan looks for this and knows it's an outside call. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] weird undocumented extensions such as s-BUSY
Andrew Kohlsmith wrote: On Tuesday 23 January 2007 1:16 pm, Barzilai Spinak wrote: So, in sum. It's just an Asterisk idiom or best(?) practice that has become somewhat common. Every time I need to do the smallest thing in Asterisk I have to Google for 3 hours and read voip-info for the more hours, and then read old mailing list post for 3 more, until I can filter out *real information* from the background noise. Maybe it's just me... I don't settle with the first solution that SeemsToWorkForNow, even though I have no idea Why or How Ah... the wonders of Early 21st Century Web Fashion and documenting everything in big lumps of Little-Ultra-Hyper-Linked-Wiki-pages with contradictory and obsolete info Hopefully it will end some day and the world will come back to its senses. After you're done hyperventilating, feel free to contribute documentation which you find is meaningful, current and insightful. Too much heat here to be hyperventilating :-) Open-source in general is very much a get your hands dirty kind of software experience. This means that you are expected to play around, experiment, and ask good questions, ALL without throwing a little tantrum as you just did. I have been getting my hands dirty for years with many kinds of OSS. What I was trying to experess by my hyperventilation is that: a) There has been a trend in the past 2-4 years of thinking that Wikiing mounts and mounts and mounts of hyperventilated (err.. linked) recipes and general babbling amounts to documentation. b) The Asterisk project in particular is worst than most OSS I've seen in this respect. Maybe aided by the fact that most people producing said documentation seem to be of the not-so technical kind and just are eager to make a quick buck by selling cheap long distance calls, so at best they just rehash someone else's recipe with a comment along the lines of this worked for me!. All this on an OSS project which has a multi-million dollar company behind it... I'm not complaining, I'm just hyperventilating some frustrations. If you want current manuals, completely stable software and someone to yell at when your system breaks, Digium offers that, too. It's called Asterisk Business Edition. I don't want the impossible current manuals as much as a current and comprehensive documentation of the architecture without all the noise from obsoleted/deprecated entities and contradictions from one wiki page to the next one touching a related concept. Otherwise, dig in, experiment and try to leave the place a little cleaner than you left it.. Cleaning... I should start with my own office :-) Believe me,. I've tried many times to comb through all voip-info.org and compiling something sensible for myself as many times I've abandoned the effort out of frustration... I don't even know where to start! It's not a matter of cleaning up, but a matter of doing it cleanly the first time. I'll keep trying though and then i'll contribute it ok.. it's finally raining now!!! done hyperventilating BarZ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM2400 Hardware Echo Cancel
Had the exact same issue with the hardware canceller. If I set echocancel=no then the problem goes away. Very weird that the static only happens on our side and we only hear it. - Original Message - From: Webster, Andrew [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, January 23, 2007 2:42 PM Subject: RE: [asterisk-users] TDM2400 Hardware Echo Cancel I have been having the same problems since installing a TDM2400 with hardware echo canceller. The best way to describe the sound is a background crackle or hiss that just can't be filtered out. Increasing the RX gain just makes the problem worse. SIP to SIP calls are flawless. An acquaintance told me the analog line level is too low, but when plugging a regular phone into the line, the signal is plenty loud enough. I am curious if anyone else had similar issues with the TDM2400 card and if they have resolved it. -- Andrew -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Adam Sharples Sent: Tuesday, January 16, 2007 09:00 To: asterisk-users@lists.digium.com Subject: [asterisk-users] TDM2400 Hardware Echo Cancel Good Day List, I'm having some issues with echo cancel on my Asterisk box, and have done extensive reading and have gained some useful pointers from this list but have a couple of hopefully fairly simple questions. The Asterisk box is connected via 20 FXO ports on a TDM2400 with the Hardware echo cancel module. Echo cancel almost works, but the users hear what they describe as a 'crackle' coming back when they talk. I want to tune to echo canceller, but am unsure if any of the options provided have any effect on the hardware module. Do the settings such as echocancel and echotraining in Zapata.conf affect the hardware module? Would I be better removing the hardware module and tuning the software echo canceller? The asterisk box is currently running 1.2.13, with zaptel 1.2. Would you advise upgrading to the newer Zaptel drivers? I don't want to upgrade Asterisk itself just yet. Any help or pointers to documentation regarding the hardware echo cancel module would be greatly appreciated, Thanks, Adam Sharples ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] OT: High Quality Wireless Headset for Cisco IPPhones and *
Plantronics 510SL Bluetooth We use this on Snom 360's in CAP positions and they work well although the lifter is a bit of a Rube Goldberg contraption. However, the receptionists are extremely happy with it, she can walk around almost the entire building with it and answer calls. One issue we found out if you have a demanding user then ergonomics come into play. One particular receptionist rejected no less than a half dozen headset solutions for ergonomic reasons. Two that stick out were (I kid you not): 1. Over-the-head headset would not be acceptable because it would mess up her hair. She has really big hair. 2. Most in-ear Bluetooth headsets she rejected because the answer button in her opinion was too small and she kept missing it. The reason? Her inch-long fingernails. Brutal. Another one she rejected was because the ear insert hurt her ear. The insert would press against the inside of her hear and irritate the multiple, chunky earrings she insisted on wearing. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Echo...
On Jan 13, 2007, at 3:38 PM, Tzafrir Cohen wrote: On Fri, Jan 12, 2007 at 04:16:55PM -0600, Matthew Fredrickson wrote: (it was loading 1.2, but you should have been loading 1.4 with the newer echo canceler). Are there impromevments in the echo canceller, or just the change of the default EC from KB1 to MG2? Well, yes, of course there are. MG2 has been changed quite a bit over it's life from 1.2 to 1.4. It's now the default because of it's now better performance heuristics. Matthew Fredrickson ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Snom 320 echo
Has anyone ever encountered an echo on the IP phone side of a call? It is an echo of the user's own voice. I believe that no one else in the office is experiencing this problem. The phone itself is a Snom 320. I've asked Snom for assistance since my source no longer carries Snom, but unlike previous times they've been slow to respond. - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Snom 320 echo
Later firmware versions have an echo-cancelling component in it, upgrade to latest version and also turn down the gains on the mic, the default setting is way too high. A setting of 3 or 4 max is all that is nessisary. hth -Original Message- From: Mike Hammett [mailto:[EMAIL PROTECTED] Sent: Tuesday, January 23, 2007 2:10 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Snom 320 echo Has anyone ever encountered an echo on the IP phone side of a call? It is an echo of the user's own voice. I believe that no one else in the office is experiencing this problem. The phone itself is a Snom 320. I've asked Snom for assistance since my source no longer carries Snom, but unlike previous times they've been slow to respond. - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com http://www.ics-il.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
AW: [asterisk-users] Snom 320 echo
Most of the cases can easily be solved by setting the handset mic gain to 2 (out of 1..8). The gain is usually much to high - optimal for whispering voices. If the other side talks loud the echo of the cable will be amplified too much. CS Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von Mike Hammett Gesendet: Dienstag, 23. Januar 2007 16:16 An: asterisk-users@lists.digium.com Betreff: [asterisk-users] Snom 320 echo Has anyone ever encountered an echo on the IP phone side of a call? It is an echo of the user's own voice. I believe that no one else in the office is experiencing this problem. The phone itself is a Snom 320. I've asked Snom for assistance since my source no longer carries Snom, but unlike previous times they've been slow to respond. - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] automon and MONITOR_EXEC
I have been trying to get Asterisk to NOT mix the in and out files from an
auto-monitored call. Something like this:
exten = s,n,SetVar(MONITOR_EXEC=/bin/true) ; do not delete files
exten = s,n,Dial(Zap/r1/${EXTEN},,wW)
Pressing *1 records as it should, but the recording is always mixed at the
end of the call.
I have figured out that the monitor is being done to the callee channel
instead of the caller channel (where MONITOR_EXEC is set). So when the
call ends, ast_monitor_stop looks for MONITOR_EXEC on the callee channel
and gets a null.
Does anyone know a workaround for this? How can I set a channel variable
on the callee channel?
~ John Williams
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[asterisk-users] DB_DELETE Function in 1.4
Does anyone know what application I should place this function in? For
example with the DB function I currently do something like this to add
an entry to the asterisk database:
exten = s,n,Set(DB(AGENT/${MACRO_EXTEN:1})=${CALLERID(num)})
To delete the entries I do something like this:
exten = s,n,DBDel(AGENT/${MACRO_EXTEN:1})
DBDel is marked as deprecated in favor of the DB_DELETE function but it
returns a warning when using it with a dialplan application like Set:
exten = s,n,Set(DB_DELETE(AGENT/${MACRO_EXTEN:1}))
Will return:
-- Executing [EMAIL PROTECTED]:202] Set(SIP/2146-b6f09f30,
DB_DELETE(AGENT/2109)) in new stack
[Jan 23 16:51:24] WARNING[4010]: pbx.c:5827 pbx_builtin_setvar: Ignoring
entry 'DB_DELETE(AGENT/2109)' with no = (and not last 'options' entry)
and it doesn't delete the database entry.
Would DB_DELETE work in an application like NoOp? Just wondering if
anyone has any experience using this new function in 1.4.0.
Thanks,
Jeremiah
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RE: [asterisk-users] * 1.0.9 Voicemail record name does not playb ack in Directory() --solved
Used the Directory application instead of the Directory AGI. -Original Message- From: Colin Anderson [mailto:[EMAIL PROTECTED] Sent: Tuesday, January 23, 2007 11:29 AM To: 'asterisk-users@lists.digium.com' Subject: [asterisk-users] * 1.0.9 Voicemail record name does not playback in Directory() On * 1.0.9 User logs into voicemail, dials Option Zero, then Option Three. Records name, accepts the recording. greet.wav is generated in the user's mailbox. It plays back fine. The Directory app still spells out his name! What am I missing? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to exit from console?
! Paul Hales Technical Manager AsteriskIT On Tue, 2007-01-23 at 22:36 +1100, Rudolf Ladyzhenskii wrote: Hi, all Stupid question, but how do you exit asterisk console without stopping the asterisk? Tried quit and exit: *CLI exit No such command 'exit' (type 'help' for help) *CLI quit No such command 'quit' (type 'help' for help) *CLI Any other ideas? I started asterisk with -cg option. Same problem if use asterisk -r to connect. Can not exit. Any ideas? Thanks, Rudolf ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem connecting PAP2 over wifi bridge
Hi All, I have my Asterisk box running with 6 extension all connected to CAT5 Grandstream phones. I'm trying to connect 2 extension on a different office across the hall by WIFI bridge using SMCWEBT-G configured as Ethernet client. If I connect the Grandstream to that box on the other office it works fine. If I connect the PAP2-NA, both extensions register with no problems with the Asterisk box but when I initiate a call it gets cut off after 20 to 30 seconds into the call but do need the PAP2 on that other office. I recorded the traffic using tcpdump and I can see that using the PAP2 the RTP traffic going to Asterisk has errors indicating: Length: 4436 (bogus, should be 340) When the full packed length is 374. SIP packets go back and forth with no problem. Any help would be really appreciated. Thank you in advance. Alfredo. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DeStar 0.2.2 released!
Hello, I'm glad to announce that DeStar 0.2.2 version has been released. This release contains a large number of bugfixes and new features, see CHANGELOG.txt for the full list. You can find it in the usual place: http://developer.berlios.de/project/showfiles.php?group_id=2112 Thanks for using DeStar, Santiago Ruano Rincón http://destar.berlios.de signature.asc Description: Esta parte del mensaje está firmada digitalmente ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [OT] Mark Spencer Presents AsteriskNOW on Youtube
Link please ? - Original Message - From: Damian Fossi [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Tuesday, January 23, 2007 9:08 PM Subject: [asterisk-users] [OT] Mark Spencer Presents AsteriskNOW on Youtube Mark Spencer, the original author and founder of Digium, presents AsteriskNOW -- Damián D. Fossi Salas ¡Software Libre hasta el 2 mil siempre! Uso: Debian Etch Kernel 2.6.18-3-686 Ubuntu Edgy Eft Kernel 2.6.15-27-amd64 Ulanix 0.4-14 Kernel 2.6.18-486 FreeBSD 6.2-RC1 Linux User: 188464 GPG Key Fingerprint = EC09 9ABA DFD8 83F0 36F3 CA89 356E 27FD E666 E6A4 Jabber ID: damianfossi en jabberes.org www.damianfossi.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Echo on IP phones...
I have a customer running Asterisk 1.2.13, Zaptel 1.2.11 with a TE110P, a TDM04B and an Astribank-32. They have been complaining that there is echo on calls even when they are IP to IP on the same network. There are 18 Aastra 9133i phones and 30 analog phones connected to the Astribank. I can understand there being a bit of echo on the analog phones, but I do not understand why there would be echo on the SIP phones when they are all using ALAW/ULAW and are on the same local network. I even have QoS configured on the Linksys SRW224P switch to give priority to the voice services. -- Telecomunicaciones Abiertas de Mexico S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [OT] Mark Spencer Presents AsteriskNOW on Youtube
On 1/23/07, Dovid B [EMAIL PROTECTED] wrote: Link please ? http://www.youtube.com/watch?v=ONOxNJquatk -- Kristian Kielhofner ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Re: [asterisk-users] How to exit from console?
Derek Whitten,hello！ *CLI stop now You can input help to see all the commands. Like this: *CLI help === 2007-01-23 23:10:12 === Rudolf Ladyzhenskii wrote: Hi, all Stupid question, but how do you exit asterisk console without stopping the asterisk? Tried quit and exit: *CLI exit No such command 'exit' (type 'help' for help) *CLI quit No such command 'quit' (type 'help' for help) *CLI Any other ideas? I started asterisk with -cg option. Same problem if use asterisk -r to connect. Can not exit. Any ideas? Thanks, Rudolf ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ctrl-c :-) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users = = = = = = = = = = = = = = = = = = = = 致 礼！ 李君 [EMAIL PROTECTED] 2007-01-24 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DB_DELETE Function in 1.4
Jeremiah Millay wrote:
Does anyone know what application I should place this function in? For
example with the DB function I currently do something like this to add
an entry to the asterisk database:
exten = s,n,Set(DB(AGENT/${MACRO_EXTEN:1})=${CALLERID(num)})
To delete the entries I do something like this:
exten = s,n,DBDel(AGENT/${MACRO_EXTEN:1})
DBDel is marked as deprecated in favor of the DB_DELETE function but it
returns a warning when using it with a dialplan application like Set:
exten = s,n,Set(DB_DELETE(AGENT/${MACRO_EXTEN:1}))
Will return:
-- Executing [EMAIL PROTECTED]:202] Set(SIP/2146-b6f09f30,
DB_DELETE(AGENT/2109)) in new stack
[Jan 23 16:51:24] WARNING[4010]: pbx.c:5827 pbx_builtin_setvar: Ignoring
entry 'DB_DELETE(AGENT/2109)' with no = (and not last 'options' entry)
and it doesn't delete the database entry.
Would DB_DELETE work in an application like NoOp? Just wondering if
anyone has any experience using this new function in 1.4.0.
Thanks,
Jeremiah
Online (CLI) reference:
*CLI core show function DB_DELETE
-= Info about function 'DB_DELETE' =-
[Syntax]
DB_DELETE(family/key)
[Synopsis]
Return a value from the database and delete it
[Description]
This function will retrieve a value from the Asterisk database
and then remove that key from the database. DB_RESULT
will be set to the key's value if it exists.
So here's what you do to delete a database entry in 1.4.0:
exten = s,n,Set(oldval=${DB_DELETE(AGENT/${MACRO_EXTEN:1})})
; saves the old value of that key (in your case the callerid)
; into ${oldval} and deletes it from the DB. You can look at
; the value for the key you just deleted.
exten = s,n,NoOp(oldval : ${oldval})
Have fun!
Alvin
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Re: [asterisk-users] realtime sipusers and rtcachefriends... bigheadache!!
hi folks, I am using asterisk 1.2.13 (debian etch). My customer's sip accounts are stored in realtime sipusers. I have enabled in sip.conf rtcachefriends=yes and ignoreregexpire=yes Each account has nat=yes Now, I have lot of problems. for example, when I change the 'secret' field of a user in the database, it doesn't get reflected in Asterisk, who is still expecting the old password. As far as I know when rtcachefriends=yes database changes are unavailable to Asterisk until a reload is performed. Cameron ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] cmd Backgound problem with option m
Hi list
I encountered problem in using Background command. Below is my
extensions.conf.
[mainmenu]
exten = 4,1,Wait(1)
exten = 4,2,Background(thank-you-for-calling)
exten = 4,3,Goto(n01|s|1)
[n01]
exten = s,1,NoOp(${CONTEXT})
exten = s,2,Background(thank-you-cooperation|m)
exten = s,3,WaitExten()
exten = s,4,Playback(digits/pound)
exten = 1,1,Playback(digits/1)
exten = i,1,Playback(digits/star)
Without m option, everything's fine.
If m option is present and when sound is playing,
- pressing 1 terminates the call and does not goto ext 1
- pressing any other key does not stop sound playing, as expected.
the message on the manager interface when 1 pressing.
-- Starting simple switch on 'Zap/1-1'
-- Executing [EMAIL PROTECTED]:1] Wait(Zap/1-1, 1) in new stack
-- Executing [EMAIL PROTECTED]:2] BackGround(Zap/1-1, thank-you-for-calling)
in new stack
-- Playing 'thank-you-for-calling' (language 'en')
-- Executing [EMAIL PROTECTED]:3] Goto(Zap/1-1, n01|s|1) in new stack
-- Goto (n01,s,1)
-- Executing [EMAIL PROTECTED]:1] NoOp(Zap/1-1, n01) in new stack
-- Executing [EMAIL PROTECTED]:2] BackGround(Zap/1-1,
thank-you-cooperation|m) in
new stack
-- Playing 'thank-you-cooperation' (language 'en')
== Spawn extension (n01, s, 2) exited non-zero on 'Zap/1-1'
-- Hungup 'Zap/1-1'
I also tried Background(thank-you-cooperation|m||n01). The result is
- pressing 1 goto ext i
- pressing any other key does not stop sound playing, as expected.
the message on the manager interface when 1 pressing.
-- Starting simple switch on 'Zap/1-1'
-- Executing [EMAIL PROTECTED]:1] Wait(Zap/1-1, 1) in new stack
-- Executing [EMAIL PROTECTED]:2] BackGround(Zap/1-1, thank-you-cooperation)
in new stack
-- Playing 'thank-you-cooperation' (language 'en')
-- Executing [EMAIL PROTECTED]:3] Goto(Zap/1-1, n01|s|1) in new stack
-- Goto (n01,s,1)
-- Executing [EMAIL PROTECTED]:1] NoOp(Zap/1-1, n01) in new stack
-- Executing [EMAIL PROTECTED]:2] BackGround(Zap/1-1,
thank-you-cooperation|m||n01)
in new stack
-- Playing 'thank-you-cooperation' (language '')
-- Sent into invalid extension 'E8' in context 'n01' on Zap/1-1
-- Executing [EMAIL PROTECTED]:1] Playback(Zap/1-1, digits/star) in new
stack
-- Playing 'digits/star' (language 'en')
== Auto fallthrough, channel 'Zap/1-1' status is 'UNKNOWN'
-- Hungup 'Zap/1-1'
NOTICE that * tries to go to ext 'E8' which is a French alphabet e with
grave accent.
DTMF detection problem? but if context option and m option of Background is
not specified, everything works well.
any help will be appreciated.
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Re: [asterisk-users] Echo on IP phones...
We have this from time to time. It's usually someone using a cheap headset that's turned up too high. Polycom's have some settings you can tweak to cancel out the echo, although they're not supported. We used them for a short while, but they seemed to interfere with the echo can on our Sangoma card, so we had to set them back to default. You might want to see if Aastra phones offer some type of internal echo can, if there are cheap headsets being used (the person not hearing the echo should turn his/her volume down), or if there's one phone in particular causing problems (could be a bad network cable or NIC on the phone). On 1/23/07, Carlos Chavez [EMAIL PROTECTED] wrote: I have a customer running Asterisk 1.2.13, Zaptel 1.2.11 with a TE110P, a TDM04B and an Astribank-32. They have been complaining that there is echo on calls even when they are IP to IP on the same network. There are 18 Aastra 9133i phones and 30 analog phones connected to the Astribank. I can understand there being a bit of echo on the analog phones, but I do not understand why there would be echo on the SIP phones when they are all using ALAW/ULAW and are on the same local network. I even have QoS configured on the Linksys SRW224P switch to give priority to the voice services. -- Telecomunicaciones Abiertas de Mexico S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] No D-channels available! Using Primary channel 16 as D-channel anyway!
Dear Gurus, i am facing some unknown problem here.. first, let me describe my case. my office is using a nortel merridian option 11c PBX. and it is connected to the provider thru a E1 card, which is working fine with no problem. my plan it to slot in a server with 2 TE11XP card to intercept the outgoing calls. Provider --te11xp--- asterisk ---te11xp-- nortel merridian option 11c i had my asterisk set up with these configuration. zaptel.conf --- loadzone=uk defaultzone=uk span=1,1,1,ccs,hdb3,crc4,yellow span=2,0,1,ccs,hdb3,crc4,yellow bchan=1-15,32-46 dchan=16,47 bchan=17-31,48-62 --- where span 1 is to the provider and span 2 is to the PBX zapata.conf - context=from-pstn switchtype=dms100 signalling=pri_cpe callerid=asreceived group=1 callgroup=1 pickupgroup=1 rxgain=0.0 txgain=0.0 channel=1-15,17-31 context=from-pstn switchtype=dms100 signalling=pri_cpe callerid=asreceived group=2 callgroup=2 pickupgroup=2 rxgain=0.0 txgain=0.0 channel=32-46,48-62 able to start asterisk. Span 2 loaded beautifully, no problem or errors, but i get this WARNING[13655]: chan_zap.c:2287 pri_find_dchan: No D-channels available! Using Primary channel 16 as D-channel anyway! warning message from span 1. which i have no idea what happened.. the funny part is, i am able to receive calls, no problem at all. CID and DID are pass thru (for incoming). but when i try to make a outgoing calls, i got errors PRI HANGUP CAUSE 1 from the provider. i did a pri debug span 1 for the call i dialed, below are the msg. - -- Accepting call from '124' to '42707898' on channel 0/21, span 2 -- Executing Set(Zap/52-1, CALLERID(number)=50399100) in new stack -- Executing NoOp(Zap/52-1, 50399100) in new stack -- Executing Dial(Zap/52-1, ZAP/g0/42707898||) in new stack -- Making new call for cr 32771 -- Requested transfer capability: 0x00 - SPEECH Protocol Discriminator: Q.931 (8) len=39 Call Ref: len= 2 (reference 3/0x3) (Originator) Message type: SETUP (5) [04 03 80 90 a3] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: A-Law (35) [18 03 a9 83 81] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 ] [6c 0a 21 80 35 30 33 39 39 31 30 30] Calling Number (len=12) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation permitted, user number not screened (0) '50399100' ] [70 09 a1 34 32 37 30 37 38 39 38] Called Number (len=11) [ Ext: 1 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '42707898' ] [a1]w*CLI Sending Complete (len= 1) -- Called g0/42707898 Protocol Discriminator: Q.931 (8) len=10 Call Ref: len= 2 (reference 3/0x3) (Terminator) Message type: CALL PROCEEDING (2) [18 03 a9 83 81] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 ] -- Processing IE 24 (cs0, Channel Identification) -- Zap/1-1 is proceeding passing it to Zap/52-1 Protocol Discriminator: Q.931 (8) len=17 Call Ref: len= 2 (reference 3/0x3) (Terminator) Message type: PROGRESS (3) [08 02 82 81] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Public network serving the local user (2) Ext: 1 Cause: Unallocated (unassigned) number (1), class = Normal Event (0) ] [1e 02 84 88] Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Public network serving the remote user (4) Ext: 1 Progress Description: Inband information or appropriate pattern now available. (8) ] [1e 02 84 82] Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Public network serving the remote user (4) Ext: 1 Progress Description: Called equipment is non-ISDN. (2) ] -- Processing IE 8 (cs0, Cause) -- Processing IE 30 (cs0, Progress Indicator) -- Processing IE 30 (cs0, Progress Indicator) -- PROGRESS with cause code 1 received -- Zap/1-1 is making progress passing it to Zap/52-1 Protocol Discriminator: Q.931 (8) len=20 Call Ref: len= 2 (reference 3/0x3) (Terminator) Message type: CONNECT (7) [1e 02 84 88] Progress Indicator (len= 4) [ Ext: 1
Re: [asterisk-users] Asterisk 1.4 Polycom buddy status
Olle E Johansson wrote: 23 jan 2007 kl. 16.09 skrev Chris Bullock: I'm running into an issue w/ Buddy status on Polycom IP650 phones using buddy status (with SIP Hints) on Asterisk 1.4. Sometimes the status on the phones will stick in the busy status. I have noticed that I can call that extension the status will reset (sometimes). Anyone else encountered this or anything similar. I've seen reports on it, but haven't been able to repeat this. I need to know a way to force this to happen, repeatably. If I can get that, I can propably trace it and fix it. It can also happen if you have packet loss in the network, of course. I've seen it happen when asterisk restarts (or possibly even just reloads SIP) without the phone being restarted - it's generally accompanied by -- Incoming call: Got SIP response 500 Internal Server Error back from 10.0.0.51 on the console. I think the status gets stuck as available most of the time, but you don't notice it because that's the default. -- James Andrewartha Systems Administrator Data Analysis Australia Pty Ltd ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No D-channels available! Using Primary channel 16 as D-channel anyway!
snip zaptel.conf --- loadzone=uk defaultzone=uk span=1,1,1,ccs,hdb3,crc4,yellow span=2,0,1,ccs,hdb3,crc4,yellow I don't think yellow alarm is necessary unless you've been advised by your carrier. bchan=1-15,32-46 dchan=16,47 bchan=17-31,48-62 --- where span 1 is to the provider and span 2 is to the PBX zapata.conf - context=from-pstn switchtype=dms100 signalling=pri_cpe callerid=asreceived group=1 callgroup=1 pickupgroup=1 rxgain=0.0 txgain=0.0 channel=1-15,17-31 context=from-pstn switchtype=dms100 signalling=pri_cpe If you are connecting the second span to the 11c, shouldn't this be pri_net? And, since you're using E1 I believe both your switchtype should be euroisdn instead of dms100. Leo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No D-channels available! Using Primary channel 16 as D-channel anyway!
i tried without yellow as well.. and according to zaptel drivers, the yellow don't do anything, just put a yellow signal where there is nothing from the provider. and yes, i did put a pri_net on the span 2, the config is a typo.. thanks for reminding me.. but still i got those errors :( Leo Ann Boon wrote: snip zaptel.conf --- loadzone=uk defaultzone=uk span=1,1,1,ccs,hdb3,crc4,yellow span=2,0,1,ccs,hdb3,crc4,yellow I don't think yellow alarm is necessary unless you've been advised by your carrier. bchan=1-15,32-46 dchan=16,47 bchan=17-31,48-62 --- where span 1 is to the provider and span 2 is to the PBX zapata.conf - context=from-pstn switchtype=dms100 signalling=pri_cpe callerid=asreceived group=1 callgroup=1 pickupgroup=1 rxgain=0.0 txgain=0.0 channel=1-15,17-31 context=from-pstn switchtype=dms100 signalling=pri_cpe If you are connecting the second span to the 11c, shouldn't this be pri_net? And, since you're using E1 I believe both your switchtype should be euroisdn instead of dms100. Leo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No D-channels available! Using Primary channel 16 as D-channel anyway!
Kong Zhen Shin wrote: i tried without yellow as well.. and according to zaptel drivers, the yellow don't do anything, just put a yellow signal where there is nothing from the provider. and yes, i did put a pri_net on the span 2, the config is a typo.. thanks for reminding me.. but still i got those errors :( Did you change your switchtype as well? Have you tried swapping the spans? Leo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No D-channels available! Using Primary channel 16 as D-channel anyway!
yes.. did that, still the same.. again, just to highlight again my problem... problem is connecting asterisk to the provider. have no problem from nortel to asterisk. i cam route all calls thru SIP. but i wan to route locals calls back to the provider.. :( this really gives me headache.. Leo Ann Boon wrote: Kong Zhen Shin wrote: i tried without yellow as well.. and according to zaptel drivers, the yellow don't do anything, just put a yellow signal where there is nothing from the provider. and yes, i did put a pri_net on the span 2, the config is a typo.. thanks for reminding me.. but still i got those errors :( Did you change your switchtype as well? Have you tried swapping the spans? Leo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
