Re: [asterisk-users] default insecure setting
If you set invite=insecure,port in the general section of sip.conf and do not mention invite settings in the user/peer section i think it will work like you want. you have to test it first coz i havent. On 2/23/07, dima [EMAIL PROTECTED] wrote: Hello, everyone. I'm having a small problem when using asterisk with GUI. For every provider I create I have to set insecure=invite,port in users.conf. Is there a way to make it a default setting? Thanks in advance. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Regards Rizwan Hisham Software Engineer ___ --Bandwidth 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] Sellvoip configuration....Please Help!!!!
hi guy, i have a problem, i have an sellvoip account and i want configure asterisk for outbound calls. this is my sip.conf register = X00:[EMAIL PROTECTED] ; this is one of the sellvoip server [sellvoip_out] type=friend secret=PassWord username=XX00 host=70.42.34.200 dtmfmode=rfc2833 context=testing disallow=all allow=ulaw extensions.conf this is a semplified context [testing] exten = 100,1,Dial(SIP/joe) exten = 101,1,Dial(SIP/andrea) exten = 110,1,Dial(SIP/joe2) exten = 611,1,Echo() whit this configuration i cant create outbound calls, i obtain this Warning message WARNING[14512]: cdr.c:509 ast_cdr_disposition: Cause not hendled Tanks for your help! ___ --Bandwidth 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 does Asterisk use SIP info command
22 feb 2007 kl. 23.40 skrev Philipp Kempgen: Olle E Johansson wrote: 22 feb 2007 kl. 19.34 skrev Philipp Kempgen: I thought it might be useful to be able to ask Asterisk for the current SIP CSeq through the Manager API in order to send your own SIP messages during a call outside of Asterisk (for AOC, whatever). Each time you ask for the CSeq Asterisk should increment the value so it does not get out of sync. Anyone sharing my opinion? We might open a feature request. We're trying to keep the Asterisk architecture multiprotocol and do things in a uniform way from the dialplan. Things like this would certainly break that, since it is very SIP- specific. Better to implement needed functionality in Asterisk. Thanks for you reply. That's basically what you have said more than once on the bug tracker. :) Thanks. Then I know that at least one person has read and understood :-) (Sorry, but sometimes it feels like being alone out there on the tracker...) Clearly SIP is not my favorite protocol as you need to go through several hundreds of pages of documentation or even more in order to implement it. And there are already too many different more or less (in)compatible implementations around. Thus I like the idea of taking a more generic approach instead of functions and applications specific to a channel driver. On the other hand people are waiting for quick solutions to blinking Snom lights and AOC without really caring for the whole picture. We do have a lot of support for blinking lamps - for devices, conferences, parking lots and now in trunk for anything. AOC is a very european thing and I keep shouting about it when I'm in Huntsville, so they're aware of the problem. There are a few patches for AOC support in the bug tracker, please review them. I know SNOM has some proprietary extensions for AOC, but what's the state on other devices? /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] Trunk version of Asterisk?
23 feb 2007 kl. 06.52 skrev Yuan LIU: Quite a few documents, including voip-info, make reference to this term. (e.g., First, You need trunk version of Asterisk.) But I can't seem to find anything that defines this. In SVN, trunk simply refers to the main body of code. Can someone explain this? You just did. trunk is the development branch, not yet released code. Not recommended for production. http://svn.digium.com/svn/asterisk/trunk/ We currently have the 1.4 release version, soon to be released in a 1.4.1 version with a lot of bugfixes. 1.2 is will soon be put in security maintenance mode, meaning we will only change it for security reasons. Releases are to be found as .tar.gz files on ftp.digium.com At this stage, I do not recommend using 1.4.0 in production - it's too buggy. Play with 1.4 from subversion, report bugs, test it - help us make sure that 1.4.1 is a good product, tested by the community. Regards, /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] peer-to-peer RTP trouble in SIP
Hey, We have asterisk 1.2.4 (old I know) with a couple of snom phones, a couple of grandstream phones and around 65 philips dect stations. Now the problem: All calls do peer to peer RTP except the calls from dect station to dect station. snom to dect or dect to snom do peer to peer. So the sip config looks fine because all the 'static deskphones' honor the REINVITE and start talking to eachother. Our supplier told us they dont send SDP with the INVITE. Can this be the problem causing dect to dect calls to always use asterisk in the RTP path ? -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called users? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Argentine Asterisk Wiki
Dear Facundo, http://www.asterisksupport.org/tiki-index.php You can create spanish pages on this tiki. Rehan Date sent: Thu, 22 Feb 2007 19:13:55 -0300 From: Facundo Ameal [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject:[asterisk-users] Argentine Asterisk Wiki Send reply to: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com asterisk-users.lists.digium.com mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] Dear Asterisk Fans, I'm an Asterisk consultant in Argentina and want to make an spanish wiki (something like voip-info.org). I have the idea and some concepts about this project. It won't be a comercial project, it would be free and it's target would be spanish talking asterisk enthusiasts. I'm wrinting these for the sake of finding contributors, people who want to help me maint this. I can manage to get a free (perhaps for a limited time) reliable hosting with the benefits of being able to install everything we want (like mediawiki, drupal, tiki-wiki or whatever) with complete access to mysql databases. Please, anyone who is interested in this send me a private e-mail. Best regards! -- 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 Rehan Ahmed Msn/Yahoo/GoogleTalk/Email: [EMAIL PROTECTED] http://www.supertec.com/ - Internet Telephony Solutions Http://www.DIDX.net - DID Number Market Place. http://blogs.rehan.com/ My Blog ~~~ First they ignore you, then they laugh at you, then they fight you, then you win. By Mahatma Gandhi. ___ --Bandwidth 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] Have an AGI script as a queue member
This is my solution to manage an Agi script as a queue member. http://www.chiese.tn.it/index.php?sezione=softwareoperazione=dettaglioid=14 The script can be simply adapted to manage may queues, many agi script. Bye nik ___ --Bandwidth 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] Help! How to get ANSWEREDTIME after DIAL a ZAP channel?
Quoting Charles Wang [EMAIL PROTECTED]: Dear Phil, The extension 'h' was a great idea although I still got the error exited non-zero. You will. Dial() always exits non-zero on hangup. -- Phil Reynolds o mail: [EMAIL PROTECTED] |L_ \ / Web: http://www.tinsleyviaduct.com/phil/ (_)- \/ Waltham 66, Emley Moor 69, Droitwich 79, Windows 95 ___ --Bandwidth 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: New tutorial: DTMF tone detection
In article [EMAIL PROTECTED], lenz [EMAIL PROTECTED] wrote: Hello list, I have prepared a small tutorial today that deals with how to avoid Asterisk rebuilding DTMF tones when using it to connect industial appliances that use DTMF. You can find it at: http://astrecipes.net/index.php?n=248 I know it isn't everybody's piece of cake, but I thought somebody could be interested as well :) Interesting idea, but wouldn't a better approach be to add a method to disable and enable the DTMF detection, either via config or dynamically in the dialplan, rather than destroy the detection capability altogether? Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Job offer near Los Angeles
I heard from a Voxbone executive that they are opening a new NOC near Los Angeles and probably hiring a few voip support engineers. If anyone is interested feel free to contact them through their website Cheers, Robert Finding fabulous fares is fun. Let Yahoo! FareChase search your favorite travel sites to find flight and hotel bargains. http://farechase.yahoo.com/promo-generic-14795097___ --Bandwidth 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 and DTMF
Hi list! I have an Asterisk server (1.2.14) connected to a E1 line via a TE410P, and some PAP2NA connected to it. The PAP2 DTMF configurations is set to INFO and Asterisk to INFO too. At first, is INFO method different from RFC2833?? Well, I have two problems. The first is that when I place a call to outside, via E1 trunk, sometimes I get some DTMF tones and I'm sure nobody hit any key. Seems like Asterisk is misinterpreting some voice frequencies as DTMF tones and is regenerating it. I think it is related to the INFO method, as Asterisk and/or PAP2 have to send it outband and the other side will generate the TONE. Is that right? Anyone experienced something like this, and have resolved it?? Ok, the second problem is that some DTMF tones I send from my phone (Connected to the PAP2) are not being interpreted by the other side of the call (generally bank systems). I had problems with it when I used inBand with G723, but now I use INFO method and still have the problem. Is my configuration right?? 00:1f.0 ISA bridge: Intel Corporation 82801EB/ER (ICH5/ICH5R) LPC Interface Brid[general] bindport=5060 bindaddr=0.0.0.0 srvlookup=yes context=default dtmfmode=info ; Do this method exists in asterisk??? Thanks! Carlos Barros ___ --Bandwidth 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] CWI, call-limit and incominglimit
Hi, In older versions of asterisk I used to be able to use incominglimit=1 to effectively disable call waiting on a specific SIP channel (Where broken phones do not allow this on the handset itself) In 1.2.x this became call-limit=1, but this prevents the phone from opening a 2nd line in order to transfer a call using attended transfer. The WiKi suggests using SetGroup() etc, but this does not cater for the case where you are Dialling several different phones simultaneously. I _could_ dial a whole bunch of Local channels, each of which checked for an extension usage count, but the additional load and complexity in the dialplan seems a bit over-the-top to me, especially when there used to be a one-line solution to this. I also considered separate user and peer sections in sip.conf, but the hosts are dynamic, and there is no way to link the IP address of the peer to the user. My best thought so far is a Macro to check each SIP entry that has CWI disabled, using SetGroup(), and removing it from the dial string if it is in use... Any better suggestions out there? Thanks, Steve ___ --Bandwidth 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() command h and H options for SIP channel
Hi, Just need to confirm whether dial() command provided options h: Allow the callee to hang up by dialing * H: Allow the caller to hang up by dialing * work for SIP channels as well ? -ag ___ --Bandwidth 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] Possible to light up a LED on Snom phones?
I use Zaptel with PRI. Can I safely install BRIStuff to get this ability
and still not break anything?
On 2/22/07, Sune Kloppenborg Jeppesen [EMAIL PROTECTED] wrote:
On Thursday 22 February 2007 23:01, Norbert Zawodsky wrote:
This sounds interesting. If it's not too complicated for you
This should get you going:
in extensions.conf:
[macro-F_Toggle_status] ; $ARG1 db family $ARG2 db key $ARG3 Device to
change
status
;http://www.voip-
info.org/wiki/index.php?page=Asterisk+cmd+BristuffDevstate
exten = s,1,Answer()
exten = s,n,Set(status=${DB(${ARG1}/${ARG2})})
exten = s,n,GotoIf($[${status} = closed]?opening|1:closing|1)
exten = opening,1,Set(status=open)
exten = opening,n,Set(DB(${ARG1}/${ARG2})=${status})
exten = opening,n,DevState(${ARG3},2)
exten = opening,n,Hangup()
exten = closing,1,Set(status=closed)
exten = closing,n,Set(DB(${ARG1}/${ARG2})=${status})
exten = closing,n,DevState(${ARG3},0)
exten = closing,n,Hangup()
Then using the value ARG3 from above:
[hint]
exten = _${ARG3},hint,DS/${ARG3}
Remember to substitute the actual variables as you can't use variables
with
hints.
Otherwise check the URL above for more info.
HTH
--
Sune Kloppenborg Jeppesen (Jaervosz)
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Re: [asterisk-users] Possible to light up a LED on Snom phones?
On 08:13, Fri 23 Feb 07, Matt wrote: I use Zaptel with PRI. Can I safely install BRIStuff to get this ability and still not break anything? Sure. bristuff will install a patched zaptel, but I never noticed it broke zaptel stuff that was working with stock zaptel. -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called users? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dial() command h and H options for SIP channel
On 18:06, Fri 23 Feb 07, ast guy wrote: Hi, Just need to confirm whether dial() command provided options h: Allow the callee to hang up by dialing * H: Allow the caller to hang up by dialing * work for SIP channels as well ? yes. It's a function in asterisk call thingie, not in the sip channel driver. -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called users? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Possible to light up a LED on Snom phones?
On Fri, 2007-02-23 at 00:25 +0100, Sune Kloppenborg Jeppesen wrote:
On Thursday 22 February 2007 23:01, Norbert Zawodsky wrote:
This sounds interesting. If it's not too complicated for you
This should get you going:
in extensions.conf:
[macro-F_Toggle_status] ; $ARG1 db family $ARG2 db key $ARG3 Device to change
status
;http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+BristuffDevstate
exten = s,1,Answer()
exten = s,n,Set(status=${DB(${ARG1}/${ARG2})})
exten = s,n,GotoIf($[${status} = closed]?opening|1:closing|1)
exten = opening,1,Set(status=open)
exten = opening,n,Set(DB(${ARG1}/${ARG2})=${status})
exten = opening,n,DevState(${ARG3},2)
exten = opening,n,Hangup()
exten = closing,1,Set(status=closed)
exten = closing,n,Set(DB(${ARG1}/${ARG2})=${status})
exten = closing,n,DevState(${ARG3},0)
exten = closing,n,Hangup()
Then using the value ARG3 from above:
[hint]
exten = _${ARG3},hint,DS/${ARG3}
Remember to substitute the actual variables as you can't use variables with
hints.
I would also insert here, that using variables in an extension name
declaration, is highly unwise!
Otherwise check the URL above for more info.
HTH
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Re: [asterisk-users] Possible to light up a LED on Snom phones?
And presumably you can roll back to the regular zaptel if need be? Does bristuff install it's OWN patched zaptel? Or do you supply the zaptel source code for it to patch? On 2/23/07, Michiel van Baak [EMAIL PROTECTED] wrote: On 08:13, Fri 23 Feb 07, Matt wrote: I use Zaptel with PRI. Can I safely install BRIStuff to get this ability and still not break anything? Sure. bristuff will install a patched zaptel, but I never noticed it broke zaptel stuff that was working with stock zaptel. -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called users? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth 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] Sellvoip configuration....Please Help!!!!
What you have in your sip.conf only handles inbound calls. You need to
add something like the following to your extensions.conf to enable
outbound calls:
Exten = _1NXXNXX, 1,
Dial(IAX2/XX00:[EMAIL PROTECTED]/${EXTEN})
Exten = _1NXXNXX, 2,
Dial(IAX2/XX00:[EMAIL PROTECTED]/${EXTEN})
Exten = _1NXXNXX, 3, Congestion()''
Please note that XX00 is your username (as assigned by
SellVOIP), and password is the password that SellVOIP assigned to your.
Good luck and have fun!
Joe
[EMAIL PROTECTED] wrote:
hi guy, i have a problem, i have an sellvoip account and i want
configure asterisk for outbound calls.
this is my sip.conf
register = X00:[EMAIL PROTECTED] ; this is one of the
sellvoip server
[sellvoip_out]
type=friend
secret=PassWord
username=XX00
host=70.42.34.200
dtmfmode=rfc2833
context=testing
disallow=all
allow=ulaw
extensions.conf
this is a semplified context
[testing]
exten = 100,1,Dial(SIP/joe)
exten = 101,1,Dial(SIP/andrea)
exten = 110,1,Dial(SIP/joe2)
exten = 611,1,Echo()
whit this configuration i cant create outbound calls, i obtain this
Warning message
WARNING[14512]: cdr.c:509 ast_cdr_disposition: Cause not hendled
Tanks for your help!
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Re: [asterisk-users] queue information into db
Hello, I'm interested too in analyzer/statistics/billing system. Can we develop together something simple? What scripts do you recomand me? Thank you, Jonson. On 2/22/07, nik600 [EMAIL PROTECTED] wrote: I am planning to develop an open source (GPL) queue statistic/analyzer. Can i use that to store data into the db? Or shall i wrote some php code to do that? On 2/22/07, lenz [EMAIL PROTECTED] wrote: Not sure about * 1.4, but you can definitely use our Qloaderd script to do that - see http://queuemetrics.com/download.jsp . That script is pretty smart (to be a loader script...) and is able to handle restarts and database disconnections. l. In data Thu, 22 Feb 2007 09:20:59 +0100, nik600 [EMAIL PROTECTED] ha scritto: Hi the new asterisk 1.4 supports to store queue log information directly into a database? (like CDR) ? thanks -- Home of QueueMetrics - http://queuemetrics.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 ___ --Bandwidth 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 information into db
In the last months i've developed a web application for the use of an asterisk call center. Yuo can - make calls from a web interface - login/logout in queue - view members logged in a queue - view callers queued in a queue - pickup a callers from a queue I am planning to add new features - queue statistic - use of ajax instead of refresh - open popup to an agent/member containing some links or information regarding the queue he is respondig Actually i use php and Asterisk Manager to do that. The biggest problem is that this application is integrated with an internal php framework that probably won't be a standard for other users. I am looking for extract only a piece of this framework to let the application work, and then start a new sf project (released under GPL). You you belive in it please help me to make a group of people interested in it, we can make some requirements analysis and start to develop an application that will be very useful for callcenter's using asterisk. Bye nik On 2/23/07, Jonson Player [EMAIL PROTECTED] wrote: Hello, I'm interested too in analyzer/statistics/billing system. Can we develop together something simple? What scripts do you recomand me? Thank you, Jonson. On 2/22/07, nik600 [EMAIL PROTECTED] wrote: I am planning to develop an open source (GPL) queue statistic/analyzer. Can i use that to store data into the db? Or shall i wrote some php code to do that? On 2/22/07, lenz [EMAIL PROTECTED] wrote: Not sure about * 1.4, but you can definitely use our Qloaderd script to do that - see http://queuemetrics.com/download.jsp . That script is pretty smart (to be a loader script...) and is able to handle restarts and database disconnections. l. In data Thu, 22 Feb 2007 09:20:59 +0100, nik600 [EMAIL PROTECTED] ha scritto: Hi the new asterisk 1.4 supports to store queue log information directly into a database? (like CDR) ? thanks -- Home of QueueMetrics - http://queuemetrics.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 ___ --Bandwidth 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] AG-188
I do believe it is that chipset. The person placing the call from the AG-188 does not hear a ring. --Mike Message: 8 Date: Fri, 23 Feb 2007 01:21:52 + From: Thomas Kenyon [EMAIL PROTECTED] Subject: Re: [asterisk-users] AG-188 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=windows-1252; format=flowed Mike Hammett wrote: Does anyone know why when calling out with an ATCOM AG-188 registered with IAX (havent tried SIP), there is no ring. Is this that you hear no ring or the other end doesn't ring? From vague memory the AG-188 is an Infineon chipset ATA (which I haven't used.) ___ --Bandwidth 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 resolve CallerID from AudioCodes FXO
Hello Angel. Did you solve this issue? I have the same problem. Thanks, José El jue, 01-02-2007 a las 01:15 -0800, Angel Heart escribió: Hi, I am using FXO 8 Ports AudioCodes. Asterisk is able to accept incoming outgoing calls. However, I noticed that the caller ID of the caller coming from the FXO displays its endpoints assigned number and not the actual caller's ID coming from PSTN. Hope someone is using the same scenario and could share on how to resolve the caller ID/Number. Thanks. Angel __ Bored stiff? Loosen up... Download and play hundreds of games for free on Yahoo! Games. ___ --Bandwidth 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] Trixbox -- ACPI and IO-APIC?
Lacy Moore - Aspendora wrote: On 2/21/07, Stephen Bosch [EMAIL PROTECTED] wrote: My point is that if it's going to involve rebuilding a kernel to support IO-APIC, then I'd just as soon build from the ground up. And my point is that this is the Asterisk Users mail list, not the Trixbox list. Either ask other there or ask on a CentOS list. I saw your point, and I disagree. Trixbox is what it is, and it is built on Asterisk. Without Asterisk, there is no Trixbox. Moreover, as long as the Trixbox forums and documentation are as weak as they are, you can expect to see Trixbox questions continue to end up on this list. People are going to keep asking Trixbox questions on this list whether you (or I) like it or not. Especially when some list members continue to answer Trixbox questions. Nothing promotes behaviour like positive reinforcement. Once you decide to build from the ground up, your Asterisk questions can be reliably answered here. Most of us don't have any idea what all kinds of weird stuff they put in Trixbox these days, which is why I saw reliably answered. The people on here could give you a solution to something that would break a Trixbox install. For the price of admission, I can hardly expect any response to any question here to be reliably answered. On this list, as in life, it is caveat emptor. Your question though, sounds like it needs to be directed to a CentOS, or as Kodak said, a RHEL list or forum. Trixbox can't be said to be a standard CentOS or RHEL release any more than it can be said to be a standard Asterisk release. A question related to kernel config is relevant here. IO-APIC directly affects whether Digium hardware works properly; it's also pretty hard to break a Trixbox install that isn't working in the first place. Finally, the Trixbox distribution is configured in a specific way; I'm not going to get a reliable answer on kernel configurations in Trixbox from anyone except someone who's used it. I know some Asterisk list members have the pure and romantic notion that this list is to be absolutely sterile and that it will tolerate no Trixbox enquiries, but real life is messier than that, especially where open source software is concerned, and cares not a whit what a purist thinks. I personally don't have any idea what you are asking, I'm pretty sure it's not an Asterisk config question, though. I don't mean to be rude, just trying to point you in the direction to get the best answers. And I don't mean to be rude, either. :) Thanks, -Stephen- ___ --Bandwidth 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] ooh323 hang up after the call is answered
Hi, I'm trying to make ooh323 works with one asterisk box running 1.2.15 version. I can ring from a h.323 to SIP and SIP to H.323, but when the call is finished when the phone is answered. This is the log when I call from the H.323 device to a SIP device: Feb 23 10:57:32 VERBOSE[6096] logger.c: -- Executing Dial(OOH323/Telconet Mantaer-c5f8, SIP/666|30|TtrwWC) in new stack Feb 23 10:57:32 DEBUG[6096] chan_sip.c: Setting NAT on RTP to 524288 Feb 23 10:57:32 DEBUG[6096] chan_sip.c: Setting NAT on VRTP to 524288 Feb 23 10:57:32 DEBUG[6096] chan_sip.c: Outgoing Call for 666 Feb 23 10:57:32 VERBOSE[6096] logger.c: -- Called 666 Feb 23 10:57:32 DEBUG[6096] channel.c: Driver for channel 'OOH323/Telconet Mantaer-c5f8' does not support indication 3, emulating it Feb 23 10:57:32 DEBUG[6096] channel.c: Prodding channel 'OOH323/Telconet Mantaer-c5f8' Feb 23 10:57:32 DEBUG[6096] channel.c: Scheduling timer at 160 sample intervals Feb 23 10:57:32 DEBUG[6079] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '[EMAIL PROTECTED]' Request 102: Found Feb 23 10:57:32 DEBUG[6079] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '[EMAIL PROTECTED]' Request 102: Found Feb 23 10:57:32 DEBUG[6068] channel.c: Avoiding initial deadlock for 'SIP/666-098cde60' Feb 23 10:57:32 VERBOSE[6096] logger.c: -- SIP/666-098cde60 is ringing Feb 23 10:57:36 DEBUG[6079] chan_sip.c: Auto destroying call '[EMAIL PROTECTED]' Feb 23 10:57:40 DEBUG[6079] chan_sip.c: Acked pending invite 102 Feb 23 10:57:40 DEBUG[6079] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Match Found Feb 23 10:57:40 DEBUG[6079] chan_sip.c: build_route: Contact hop: Guillermo Salas M sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp Feb 23 10:57:40 VERBOSE[6096] logger.c: -- SIP/666-098cde60 answered OOH323/Telconet Mantaer-c5f8 Feb 23 10:57:40 WARNING[6096] src/chan_h323.c: Don't know how to indicate condition -1 on ooh323c_1 Feb 23 10:57:40 DEBUG[6096] channel.c: Scheduling timer at 0 sample intervals Feb 23 10:57:40 DEBUG[6096] channel.c: Didn't get a frame from channel: OOH323/Telconet Mantaer-c5f8 Feb 23 10:57:40 DEBUG[6096] channel.c: Bridge stops bridging channels OOH323/Telconet Mantaer-c5f8 and SIP/666-098cde60 Feb 23 10:57:40 DEBUG[6096] chan_sip.c: update_call_counter(666) - decrement call limit counter Feb 23 10:57:40 DEBUG[6096] app_dial.c: Exiting with DIALSTATUS=ANSWER. Feb 23 10:57:40 VERBOSE[6096] logger.c: == Spawn extension (macro-dial, s, 10) exited non-zero on 'OOH323/Telconet Mantaer-c5f8' in macro 'dial' Feb 23 10:57:40 VERBOSE[6096] logger.c: == Spawn extension (macro-dial, s, 10) exited non-zero on 'OOH323/Telconet Mantaer-c5f8' in macro 'exten-vm' Feb 23 10:57:40 VERBOSE[6096] logger.c: == Spawn extension (macro-dial, s, 10) exited non-zero on 'OOH323/Telconet Mantaer-c5f8' And the h323 log: 10:57:32:717 Created a new call (incoming, ooh323c_1) 10:57:32:753 Received SETUP message (incoming, ooh323c_1) 10:57:32:753 Tunneling disabled by remote endpoint. (incoming, ooh323c_1) 10:57:32:753 Enabled RFC2833 DTMF capability for (incoming, ooh323c_1) 10:57:32:754 Sent Message - CallProceeding (incoming, ooh323c_1) 10:57:32:754 Sent Message - Alerting (incoming, ooh323c_1) 10:57:40:475 Cmd connection accepted 10:57:40:476 Processing Answer Call command for ooh323c_1 10:57:40:476 Creating H245 listener 10:57:40:476 H245 listener creation - successful(port 12031) (incoming, ooh323c_1) 10:57:40:476 H.245 Listerner socket being monitored (incoming, ooh323c_1) 10:57:40:476 Sent Message - Connect (incoming, ooh323c_1) 10:57:40:476 H.245 Listerner socket being monitored (incoming, ooh323c_1) 10:57:40:501 H.245 connection established (incoming, ooh323c_1) 10:57:40:501 Sent Message - TerminalCapabilitySet (incoming, ooh323c_1) 10:57:40:502 Sent Message - MasterSlaveDetermination (incoming, ooh323c_1) 10:57:40:538 Sent Message - TerminalCapabilitySetAck (incoming, ooh323c_1) 10:57:40:542 Master Slave Determination received (incoming, ooh323c_1) 10:57:40:542 MasterSlaveDetermination done - Slave(incoming, ooh323c_1) 10:57:40:542 Sent Message - MasterSlaveDeterminationAck (incoming, ooh323c_1) 10:57:40:556 Opening logical channels (incoming, ooh323c_1) 10:57:40:556 ERROR:Local endpoint does not have any audio capabilities (incoming, ooh323c_1) 10:57:40:556 ERROR:Failed to open audio channels. Clearing call.(incoming, ooh323c_1) 10:57:40:556 Sent Message - EndSessionCommand (incoming, ooh323c_1) 10:57:40:556 Sent Message - ReleaseComplete (incoming, ooh323c_1) 10:57:40:562 Received EndSession command (incoming, ooh323c_1) 10:57:40:562 Closing H.245 connection (incoming, ooh323c_1) 10:57:40:562 H.245 Listerner socket being monitored (incoming, ooh323c_1) 10:57:40:577 H.225 Release Complete message received (incoming, ooh323c_1) 10:57:40:577 Release complete reason code 12. (incoming, ooh323c_1) 10:57:40:577 Cleaning Call (incoming,
RE: [asterisk-users] upgrading from A101 to....A102
We're having a lot of D channel problems with the pci-e on HP servers. Going to PCI fixed the problem. Sangoma is aware of the problem and is using one of our servers to work toward a solution. -Jeremy From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bill Gibbs Sent: Thursday, February 22, 2007 2:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] upgrading from A101 toA102 Any benefit on getting the PCI Express version? Bill ___ --Bandwidth 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] ooh323 hang up after the call is answered
I fogot, the H.323 device is one Antek networks INC with two fxo ports. Regards, On Fri, 2007-02-23 at 11:07 -0500, Guillermo Salas M. wrote: Hi, I'm trying to make ooh323 works with one asterisk box running 1.2.15 version. I can ring from a h.323 to SIP and SIP to H.323, but when the call is finished when the phone is answered. This is the log when I call from the H.323 device to a SIP device: Feb 23 10:57:32 VERBOSE[6096] logger.c: -- Executing Dial(OOH323/Telconet Mantaer-c5f8, SIP/666|30|TtrwWC) in new stack Feb 23 10:57:32 DEBUG[6096] chan_sip.c: Setting NAT on RTP to 524288 Feb 23 10:57:32 DEBUG[6096] chan_sip.c: Setting NAT on VRTP to 524288 Feb 23 10:57:32 DEBUG[6096] chan_sip.c: Outgoing Call for 666 Feb 23 10:57:32 VERBOSE[6096] logger.c: -- Called 666 Feb 23 10:57:32 DEBUG[6096] channel.c: Driver for channel 'OOH323/Telconet Mantaer-c5f8' does not support indication 3, emulating it Feb 23 10:57:32 DEBUG[6096] channel.c: Prodding channel 'OOH323/Telconet Mantaer-c5f8' Feb 23 10:57:32 DEBUG[6096] channel.c: Scheduling timer at 160 sample intervals Feb 23 10:57:32 DEBUG[6079] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '[EMAIL PROTECTED]' Request 102: Found Feb 23 10:57:32 DEBUG[6079] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '[EMAIL PROTECTED]' Request 102: Found Feb 23 10:57:32 DEBUG[6068] channel.c: Avoiding initial deadlock for 'SIP/666-098cde60' Feb 23 10:57:32 VERBOSE[6096] logger.c: -- SIP/666-098cde60 is ringing Feb 23 10:57:36 DEBUG[6079] chan_sip.c: Auto destroying call '[EMAIL PROTECTED]' Feb 23 10:57:40 DEBUG[6079] chan_sip.c: Acked pending invite 102 Feb 23 10:57:40 DEBUG[6079] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Match Found Feb 23 10:57:40 DEBUG[6079] chan_sip.c: build_route: Contact hop: Guillermo Salas M sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp Feb 23 10:57:40 VERBOSE[6096] logger.c: -- SIP/666-098cde60 answered OOH323/Telconet Mantaer-c5f8 Feb 23 10:57:40 WARNING[6096] src/chan_h323.c: Don't know how to indicate condition -1 on ooh323c_1 Feb 23 10:57:40 DEBUG[6096] channel.c: Scheduling timer at 0 sample intervals Feb 23 10:57:40 DEBUG[6096] channel.c: Didn't get a frame from channel: OOH323/Telconet Mantaer-c5f8 Feb 23 10:57:40 DEBUG[6096] channel.c: Bridge stops bridging channels OOH323/Telconet Mantaer-c5f8 and SIP/666-098cde60 Feb 23 10:57:40 DEBUG[6096] chan_sip.c: update_call_counter(666) - decrement call limit counter Feb 23 10:57:40 DEBUG[6096] app_dial.c: Exiting with DIALSTATUS=ANSWER. Feb 23 10:57:40 VERBOSE[6096] logger.c: == Spawn extension (macro-dial, s, 10) exited non-zero on 'OOH323/Telconet Mantaer-c5f8' in macro 'dial' Feb 23 10:57:40 VERBOSE[6096] logger.c: == Spawn extension (macro-dial, s, 10) exited non-zero on 'OOH323/Telconet Mantaer-c5f8' in macro 'exten-vm' Feb 23 10:57:40 VERBOSE[6096] logger.c: == Spawn extension (macro-dial, s, 10) exited non-zero on 'OOH323/Telconet Mantaer-c5f8' And the h323 log: 10:57:32:717 Created a new call (incoming, ooh323c_1) 10:57:32:753 Received SETUP message (incoming, ooh323c_1) 10:57:32:753 Tunneling disabled by remote endpoint. (incoming, ooh323c_1) 10:57:32:753 Enabled RFC2833 DTMF capability for (incoming, ooh323c_1) 10:57:32:754 Sent Message - CallProceeding (incoming, ooh323c_1) 10:57:32:754 Sent Message - Alerting (incoming, ooh323c_1) 10:57:40:475 Cmd connection accepted 10:57:40:476 Processing Answer Call command for ooh323c_1 10:57:40:476 Creating H245 listener 10:57:40:476 H245 listener creation - successful(port 12031) (incoming, ooh323c_1) 10:57:40:476 H.245 Listerner socket being monitored (incoming, ooh323c_1) 10:57:40:476 Sent Message - Connect (incoming, ooh323c_1) 10:57:40:476 H.245 Listerner socket being monitored (incoming, ooh323c_1) 10:57:40:501 H.245 connection established (incoming, ooh323c_1) 10:57:40:501 Sent Message - TerminalCapabilitySet (incoming, ooh323c_1) 10:57:40:502 Sent Message - MasterSlaveDetermination (incoming, ooh323c_1) 10:57:40:538 Sent Message - TerminalCapabilitySetAck (incoming, ooh323c_1) 10:57:40:542 Master Slave Determination received (incoming, ooh323c_1) 10:57:40:542 MasterSlaveDetermination done - Slave(incoming, ooh323c_1) 10:57:40:542 Sent Message - MasterSlaveDeterminationAck (incoming, ooh323c_1) 10:57:40:556 Opening logical channels (incoming, ooh323c_1) 10:57:40:556 ERROR:Local endpoint does not have any audio capabilities (incoming, ooh323c_1) 10:57:40:556 ERROR:Failed to open audio channels. Clearing call.(incoming, ooh323c_1) 10:57:40:556 Sent Message - EndSessionCommand (incoming, ooh323c_1) 10:57:40:556 Sent Message - ReleaseComplete (incoming, ooh323c_1) 10:57:40:562 Received EndSession command (incoming, ooh323c_1) 10:57:40:562 Closing H.245 connection (incoming, ooh323c_1) 10:57:40:562
[asterisk-users] SLA more than 100% ?
How does one answer more than 100% of the calls in less than 60 seconds? techsupp has 0 calls (max 20) in 'leastrecent' strategy (104s holdtime), W:0, C:3, A:2, SL:166.7% within 60s ___ --Bandwidth 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] Trixbox -- ACPI and IO-APIC?
On 2/23/07, Stephen Bosch [EMAIL PROTECTED] wrote: I saw your point, and I disagree. Trixbox is what it is, and it is built on Asterisk. Without Asterisk, there is no Trixbox. Moreover, as long as the Trixbox forums and documentation are as weak as they are, you can expect to see Trixbox questions continue to end up on this list. People are going to keep asking Trixbox questions on this list whether you (or I) like it or not. Especially when some list members continue to answer Trixbox questions. Nothing promotes behaviour like positive reinforcement. Stephen, Thank you for saying this. I wish more Trixbox users and developers would remember that trixbox is based off of Asterisk and without Asterisk, there would be no trixbox. Most of the negative attitude towards Trixbox in the Asterisk community stems from the fact that Trixbox/Fonality LOVES to brag about their community - the biggest Asterisk community, forum posts, forum members, features, etc. At one point they bragged that they have more users than Digium. How they think they can claim this I have no idea. Anyways, after that rant I'll try to answer your question... Originally APIC was used solely for SMP systems. Most newer motherboards (even if they are uniprocessor) now have an APIC (finally). However, the Linux kernel allows you to disable APIC controllers on uniprocessor motherboards. This is the default for every RedHat derived kernel that I know of. This obviously includes CentOS and trixbox. On non-SMP kernels there is a KCONFIG option for this - CONFIG_X86_UP_APIC. So, to answer your question... Is your system SMP? If so, does it have an SMP kernel? If it isn't SMP and you want to use APIC on your system, you will probably have to recompile your kernel with CONFIG_X86_UP_APIC enabled. Shameless plug - If you want a distro that was designed from the ground up for Asterisk and Zaptel cards, take a look at AstLinux - http://www.astlinux.org. Your APIC will certainly work ;). -- 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
[asterisk-users] Polycom SIP 501 Transfer Question
I know this is not a Polycom support forum, but I also know there are a lot of you with a great deal of Polycom experience. Is there anyway to remove the Attended Transfer but keep the Blind transfer? Or better yet, just swap the two soft buttons locations? I know you can remap the Hard buttons, but what about the soft buttons? The reason I need this is my users can't get it through their head that they need to announce the call if they use the normal aka Attended transfer before the press the transfer button again to complete it. I know if they would just use the Blind transfer we would have no problems, but since the Blind transfer is on the second set of screen soft buttons they aren't smart enough to find it I guess. The problem with them using Attended Transfer is CallerID shows up as theirs, when in reality they have already press the transfer button a second time. We then don't answer the phone professionally since we think that it is our employee calling us. Thanks! --Shawn ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom SIP 501 Transfer Question
In later 1.6.x firmwares there is a config option for allow transfer on proceeding that basically allows you to do a blind transfer by just hitting the transfer key again rather than having to select Blind. Shawn Kelley wrote: I know this is not a Polycom support forum, but I also know there are a lot of you with a great deal of Polycom experience. Is there anyway to remove the Attended Transfer but keep the Blind transfer? Or better yet, just swap the two soft buttons locations? I know you can remap the Hard buttons, but what about the soft buttons? The reason I need this is my users can't get it through their head that they need to announce the call if they use the normal aka Attended transfer before the press the transfer button again to complete it. I know if they would just use the Blind transfer we would have no problems, but since the Blind transfer is on the second set of screen soft buttons they aren't smart enough to find it I guess. The problem with them using Attended Transfer is CallerID shows up as theirs, when in reality they have already press the transfer button a second time. We then don't answer the phone professionally since we think that it is our employee calling us. ___ --Bandwidth 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 does Asterisk use SIP info command
Olle E Johansson wrote: 22 feb 2007 kl. 23.40 skrev Philipp Kempgen: Olle E Johansson wrote: 22 feb 2007 kl. 19.34 skrev Philipp Kempgen: I thought it might be useful to be able to ask Asterisk for the current SIP CSeq through the Manager API in order to send your own SIP messages during a call outside of Asterisk (for AOC, whatever). Each time you ask for the CSeq Asterisk should increment the value so it does not get out of sync. Anyone sharing my opinion? We might open a feature request. We're trying to keep the Asterisk architecture multiprotocol and do things in a uniform way from the dialplan. Things like this would certainly break that, since it is very SIP- specific. Better to implement needed functionality in Asterisk. Thanks for you reply. That's basically what you have said more than once on the bug tracker. :) Thanks. Then I know that at least one person has read and understood :-) (Sorry, but sometimes it feels like being alone out there on the tracker...) :) On the other hand people are waiting for quick solutions to blinking Snom lights and AOC without really caring for the whole picture. We do have a lot of support for blinking lamps - for devices, conferences, parking lots and now in trunk for anything. People refrain from using the trunk in a production environment. And as I can remember even the trunk does not address the Snom pickup problem. That's on of the things bristuff is popular for. AOC is a very european thing and I keep shouting about it when I'm in Huntsville, so they're aware of the problem. Great. :) For a european company it's like this: We have AOC now, can we have that with Asterisk? No. (Or at least not very easily, eg. without a patch) But you probably know that. There are a few patches for AOC support in the bug tracker, please review them. I know SNOM has some proprietary extensions for AOC, but what's the state on other devices? Snom has this page in their Wiki: http://www.snom.com/wiki/index.php/Advice_of_charge_(AOC)_in_SIP But they don't really say whether they actually use this in their phones or if it's more like a working draft. Apart from http://tools.ietf.org/html/draft-garcia-sipping-etsi-ngn-p-headers-00#section-4.1 is there any other standard that I should be aware of? Regards, Philipp -- 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 Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --Bandwidth 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 and Cisco PRI gateway config
Hi, Do you think it could have been done with another T1/E1Asterisk box between the Nortel PBX and the other Asterisk server ? Which features would you then loose or gain, given current status of QSIG support in Asterisk ? Regards ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP Test
Hi, I have just setup inbound SIP and wonder if somebody would be so kind as to test that it works for me, and that my firewall is setup okay. sip:[EMAIL PROTECTED] Thank you -- --[ UxBoD ]-- // PGP Key: curl -s http://www.splatnix.net/uxbod.asc | gpg --import // Fingerprint: 543A E778 7F2D 98F1 3E50 9C1F F190 93E0 E8E8 0CF8 // Keyserver: www.keyserver.net Key-ID: 0xE8E80CF8 -- 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] Polycom SIP 501 Transfer Question
Yeah but I think the caller ID issue still remains. Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric ManxPower Wieling Sent: Friday, February 23, 2007 11:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom SIP 501 Transfer Question In later 1.6.x firmwares there is a config option for allow transfer on proceeding that basically allows you to do a blind transfer by just hitting the transfer key again rather than having to select Blind. Shawn Kelley wrote: I know this is not a Polycom support forum, but I also know there are a lot of you with a great deal of Polycom experience. Is there anyway to remove the Attended Transfer but keep the Blind transfer? Or better yet, just swap the two soft buttons locations? I know you can remap the Hard buttons, but what about the soft buttons? The reason I need this is my users can't get it through their head that they need to announce the call if they use the normal aka Attended transfer before the press the transfer button again to complete it. I know if they would just use the Blind transfer we would have no problems, but since the Blind transfer is on the second set of screen soft buttons they aren't smart enough to find it I guess. The problem with them using Attended Transfer is CallerID shows up as theirs, when in reality they have already press the transfer button a second time. We then don't answer the phone professionally since we think that it is our employee calling us. ___ --Bandwidth 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] ooh323 hang up after the call is answered
Solved... installed chan_oh323 :) http://www.ecualug.org/?q=2006/02/28/comos/asterisk_1_2_4_agregando_soporte_para_el_protocolo_h_323 I don't know why ooh323 does not work. Regards, On Fri, 2007-02-23 at 11:21 -0500, Guillermo Salas M. wrote: I fogot, the H.323 device is one Antek networks INC with two fxo ports. Regards, On Fri, 2007-02-23 at 11:07 -0500, Guillermo Salas M. wrote: Hi, I'm trying to make ooh323 works with one asterisk box running 1.2.15 version. I can ring from a h.323 to SIP and SIP to H.323, but when the call is finished when the phone is answered. This is the log when I call from the H.323 device to a SIP device: Feb 23 10:57:32 VERBOSE[6096] logger.c: -- Executing Dial(OOH323/Telconet Mantaer-c5f8, SIP/666|30|TtrwWC) in new stack Feb 23 10:57:32 DEBUG[6096] chan_sip.c: Setting NAT on RTP to 524288 Feb 23 10:57:32 DEBUG[6096] chan_sip.c: Setting NAT on VRTP to 524288 Feb 23 10:57:32 DEBUG[6096] chan_sip.c: Outgoing Call for 666 Feb 23 10:57:32 VERBOSE[6096] logger.c: -- Called 666 Feb 23 10:57:32 DEBUG[6096] channel.c: Driver for channel 'OOH323/Telconet Mantaer-c5f8' does not support indication 3, emulating it Feb 23 10:57:32 DEBUG[6096] channel.c: Prodding channel 'OOH323/Telconet Mantaer-c5f8' Feb 23 10:57:32 DEBUG[6096] channel.c: Scheduling timer at 160 sample intervals Feb 23 10:57:32 DEBUG[6079] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '[EMAIL PROTECTED]' Request 102: Found Feb 23 10:57:32 DEBUG[6079] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '[EMAIL PROTECTED]' Request 102: Found Feb 23 10:57:32 DEBUG[6068] channel.c: Avoiding initial deadlock for 'SIP/666-098cde60' Feb 23 10:57:32 VERBOSE[6096] logger.c: -- SIP/666-098cde60 is ringing Feb 23 10:57:36 DEBUG[6079] chan_sip.c: Auto destroying call '[EMAIL PROTECTED]' Feb 23 10:57:40 DEBUG[6079] chan_sip.c: Acked pending invite 102 Feb 23 10:57:40 DEBUG[6079] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Match Found Feb 23 10:57:40 DEBUG[6079] chan_sip.c: build_route: Contact hop: Guillermo Salas M sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp Feb 23 10:57:40 VERBOSE[6096] logger.c: -- SIP/666-098cde60 answered OOH323/Telconet Mantaer-c5f8 Feb 23 10:57:40 WARNING[6096] src/chan_h323.c: Don't know how to indicate condition -1 on ooh323c_1 Feb 23 10:57:40 DEBUG[6096] channel.c: Scheduling timer at 0 sample intervals Feb 23 10:57:40 DEBUG[6096] channel.c: Didn't get a frame from channel: OOH323/Telconet Mantaer-c5f8 Feb 23 10:57:40 DEBUG[6096] channel.c: Bridge stops bridging channels OOH323/Telconet Mantaer-c5f8 and SIP/666-098cde60 Feb 23 10:57:40 DEBUG[6096] chan_sip.c: update_call_counter(666) - decrement call limit counter Feb 23 10:57:40 DEBUG[6096] app_dial.c: Exiting with DIALSTATUS=ANSWER. Feb 23 10:57:40 VERBOSE[6096] logger.c: == Spawn extension (macro-dial, s, 10) exited non-zero on 'OOH323/Telconet Mantaer-c5f8' in macro 'dial' Feb 23 10:57:40 VERBOSE[6096] logger.c: == Spawn extension (macro-dial, s, 10) exited non-zero on 'OOH323/Telconet Mantaer-c5f8' in macro 'exten-vm' Feb 23 10:57:40 VERBOSE[6096] logger.c: == Spawn extension (macro-dial, s, 10) exited non-zero on 'OOH323/Telconet Mantaer-c5f8' And the h323 log: 10:57:32:717 Created a new call (incoming, ooh323c_1) 10:57:32:753 Received SETUP message (incoming, ooh323c_1) 10:57:32:753 Tunneling disabled by remote endpoint. (incoming, ooh323c_1) 10:57:32:753 Enabled RFC2833 DTMF capability for (incoming, ooh323c_1) 10:57:32:754 Sent Message - CallProceeding (incoming, ooh323c_1) 10:57:32:754 Sent Message - Alerting (incoming, ooh323c_1) 10:57:40:475 Cmd connection accepted 10:57:40:476 Processing Answer Call command for ooh323c_1 10:57:40:476 Creating H245 listener 10:57:40:476 H245 listener creation - successful(port 12031) (incoming, ooh323c_1) 10:57:40:476 H.245 Listerner socket being monitored (incoming, ooh323c_1) 10:57:40:476 Sent Message - Connect (incoming, ooh323c_1) 10:57:40:476 H.245 Listerner socket being monitored (incoming, ooh323c_1) 10:57:40:501 H.245 connection established (incoming, ooh323c_1) 10:57:40:501 Sent Message - TerminalCapabilitySet (incoming, ooh323c_1) 10:57:40:502 Sent Message - MasterSlaveDetermination (incoming, ooh323c_1) 10:57:40:538 Sent Message - TerminalCapabilitySetAck (incoming, ooh323c_1) 10:57:40:542 Master Slave Determination received (incoming, ooh323c_1) 10:57:40:542 MasterSlaveDetermination done - Slave(incoming, ooh323c_1) 10:57:40:542 Sent Message - MasterSlaveDeterminationAck (incoming, ooh323c_1) 10:57:40:556 Opening logical channels (incoming, ooh323c_1) 10:57:40:556 ERROR:Local endpoint does not have any audio capabilities (incoming, ooh323c_1) 10:57:40:556
[asterisk-users] Re: How to separate outgoing extens from the contexts from sip.conf?
Problem solved and posted below. I may have found out at lease two things that may help others. Especial thanks to Eric ManPower, Benny Amorsen, Luan LIU, Paul Hales, Pavel Jezek and a _lot_ of research on the web. The problem was separating the contexts for incoming VOIP calls from the outgoing trunks to eliminate the possibility of an outside caller being able to make calls (international, 900 numbers etc) on my dime. This _might_ be a new wrinkle but I believe it should apply to most setups that use a 'gateway' SIP trunk to access their ISP. My setup has gateway SIP trunks to my provider Telasip.com and my Sipura SPA-3000 for PSTN calls. Sipura is easy to separate since the Voxilla Sipura setup wizard setup separate inbound FXO to CO and outbound trunks FXS from POTS phones. Most ISP trunks would be expected to be a Peer and thus have no context. Telasip instructs their customers to setup the gateway as a Friend with a corresponding context=telasip-incoming line in sip.conf. This makes it easy to process all incoming calls from Telasip completely separate from dialed calls which are handled separately in the sip phones context=sip-outgoing. Since my system is for a single family, all users have full access to long distance (no teenagers!) although it would be easy to have classes of sip phones with different privileges. Telasip told me that it was necessary to have a line in their gateway sip.conf entry that 'insecure=very'. I have found Telasip to be a most excellent provider with great support. The other thing is that I found out on the voip-info wiki that a [default] context in extensions.conf is treated different from other contexts. If there is no context=something line in the [general] sip.conf, then all calls that cannot find an extension to goto will come into [default] or [something] if such a line exists in sip.conf [general]. I didn't know _that_. Voip-info warns against putting any lines in [default] that might allow a caller to call out in that context. In my case, anyone who manages to get into [default] goes direct to voicemail and, when they finish with that, get either a hangup or congestion to dispose of the call. Of course, if they wish to leave a vm I'll consider it but I expect that few will. There is nothing else in [default]. Finally about 900 calls. My 10/11 dial plan lines exclude 900 calls. My wife and I don't need them any more vbg Hope this information will be helpful to someone else. Larry -- Larry Alkoff N2LA - Austin TX Using Thunderbird on Linux ___ --Bandwidth 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] cisco sip firmware update for cisco 7970
I'm trying to buy the cisco firmware update but it seems that i cannot order online because I bought my 7970 on ebay. Is there any other chance to get this update? ... anyone can make me a favour and send it to me by email? thank you ___ --Bandwidth 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] SOLVED: Call forwarding and 1.2.x
We had an issue, and I know others had posted the same on the list. Scenario: Polycom phone user sets call forward to a toll free number(in our case) Call arrives for the phone, the phone notifys asterisk, asterisk dials new number. Telco drops call. But if you dial direct to the number it is a good working number. Solution Turns out our carriers DMS had a tuple on the PRI set incorrect. Seems they did not like the call forward information element sent in rn format. Setting the tuple correctly solved the issue. But it took the carrier a call into Nortel to have them figure it out. Switch techs had never seen that tuple used before. Still not sure what the rn format vs any other is yet. Anyone know? ___ --Bandwidth 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] Queue Macro Problem
Hey all,
This should be an easy one. I have a few different queues and wanted to
set up a standard macro to handle them, so I can shrink the dial plan
down and stop having so much redundancy. But when I try to use it, i get
a no answer.
Here's what does work (non macro):
exten = 5054,1,Answer()
exten = 5054,n,Ringing()
exten = 5054,n,Wait(2)
exten = 5054,n,Queue(itdept,t|||30)
exten = 5054,n,Voicemail(u5054) ; If unavailable, send to voicemail
exten = 5054,n,Hangup
Here's the macro I tried to make and use:
[coqueuevm]
; Call One Queue - Goto to voicemail after 30 secs
; ${ARG1} - Queue Name
; ${ARG2} - Voicemail
exten = s,1,Answer()
exten = s,n,Ringing()
exten = s,n,Wait(2)
exten = s,n,Queue(${ARG1},t|||30)
exten = s,n,Voicemail(u${ARG2})
exten = s,n,Hangup()
--
exten = 5054,1,Macro(coqueuevm,itdept,5054)
any thoughts?
Rob
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Re: [asterisk-users] SIP Test
On Fri, 23 Feb 2007 19:09:40 + --[ UxBoD ]-- [EMAIL PROTECTED] wrote: Hi, I have just setup inbound SIP and wonder if somebody would be so kind as to test that it works for me, and that my firewall is setup okay. sip:[EMAIL PROTECTED] Thank you Thank you very much to westcomuk for leaving the message, and to others who have tested for me :) A question though is that is shows the respondent as sip:@my sip server and not the actual originating caller. Why would that be ? Thanks, -- --[ UxBoD ]-- // PGP Key: curl -s http://www.splatnix.net/uxbod.asc | gpg --import // Fingerprint: 543A E778 7F2D 98F1 3E50 9C1F F190 93E0 E8E8 0CF8 // Keyserver: www.keyserver.net Key-ID: 0xE8E80CF8 // SIP:[EMAIL PROTECTED] -- 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
[asterisk-users] asterisk
Hi i install Asterisk can register softphones on clients computers but when i make a call to a extencion this error apear Call Failed: not found in the asterisk machine i do commannd sip show peers and i can see the clients there can you help me thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue Macro Problem
Rob Schall wrote: Here's the macro I tried to make and use: [coqueuevm] The name of the context should be macro-coqueuevm. The macro- part will automatically be cut by the Macro() application. exten = 5054,1,Macro(coqueuevm,itdept,5054) Regards, Philipp -- 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 Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --Bandwidth 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 Macro Problem
That was it. :) Thanks much! A followup... well, kinda related... And not really a asterisk q. On the polycom 501 phones... There's those 3 lines that you can setup. Is it possible to make one of them a shortcut to the queue login/logout extension? Rob Philipp Kempgen wrote: Rob Schall wrote: Here's the macro I tried to make and use: [coqueuevm] The name of the context should be macro-coqueuevm. The macro- part will automatically be cut by the Macro() application. exten = 5054,1,Macro(coqueuevm,itdept,5054) Regards, Philipp ___ --Bandwidth 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 Macro Problem
Rob Schall wrote: On the polycom 501 phones... There's those 3 lines that you can setup. Is it possible to make one of them a shortcut to the queue login/logout extension? Haven't used that phone myself but it seems like you need to add your queue extension (5054) to the phone's directory and assign one of the unused line keys to that directory entry. Regards, Philipp -- 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 Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --Bandwidth 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 Macro Problem
Philipp Kempgen wrote: Rob Schall wrote: On the polycom 501 phones... There's those 3 lines that you can setup. Is it possible to make one of them a shortcut to the queue login/logout extension? Haven't used that phone myself but it seems like you need to add your queue extension (5054) to the phone's directory Mea culpa. 5054 is not your login/logout extension. You know what I mean. :) and assign one of the unused line keys to that directory entry. Regards, Philipp -- 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 Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --Bandwidth 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] GSM cleanup (pops, clicks and static)
I have a bunch of sounds that I've converted into gsm from (Indexed NMS) vox files. There's only a single utility that I've found that can read and convert vox files. My conversion process is to use this utility to convert the index vox file in to a series of wave files and then use sox to convert the wave files to gsm files. Over all this works really well, the problem is that about 60 to 70 percent of the gsm files have some static or popping and clicking, on most of them it is in the silence at the end of the file. All that back story to ask this question: Are there any good utilities available for cleaning up gsm files? Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] GSM cleanup (pops, clicks and static)
Audacity -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Ruggles Sent: Friday, February 23, 2007 4:48 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] GSM cleanup (pops, clicks and static) I have a bunch of sounds that I've converted into gsm from (Indexed NMS) vox files. There's only a single utility that I've found that can read and convert vox files. My conversion process is to use this utility to convert the index vox file in to a series of wave files and then use sox to convert the wave files to gsm files. Over all this works really well, the problem is that about 60 to 70 percent of the gsm files have some static or popping and clicking, on most of them it is in the silence at the end of the file. All that back story to ask this question: Are there any good utilities available for cleaning up gsm files? Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network Engineer Safe Data, Inc. (910) 285-7200[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 ___ --Bandwidth 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] GSM cleanup (pops, clicks and static)
On Fri, 2007-02-23 at 16:48 -0500, David Ruggles wrote: I have a bunch of sounds that I've converted into gsm from (Indexed NMS) vox files. There's only a single utility that I've found that can read and convert vox files. My conversion process is to use this utility to convert the index vox file in to a series of wave files and then use sox to convert the wave files to gsm files. Over all this works really well, the problem is that about 60 to 70 percent of the gsm files have some static or popping and clicking, on most of them it is in the silence at the end of the file. All that back story to ask this question: Are there any good utilities available for cleaning up gsm files? I've been playing around quite a bit lately! If I were you, I'd take those wav files, and use audacity to inspect and clean them, then do the sox thing to get to gsm. If they are clean in wav, they should be clean in gsm. murf -- Steve Murphy Software Developer Digium 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
Re: [asterisk-users] GSM cleanup (pops, clicks and static)
Wavepad ( a windows program ) is MUCH easier to use John Novack Robert Augustyn wrote: Audacity -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Ruggles Sent: Friday, February 23, 2007 4:48 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] GSM cleanup (pops, clicks and static) I have a bunch of sounds that I've converted into gsm from (Indexed NMS) vox files. There's only a single utility that I've found that can read and convert vox files. My conversion process is to use this utility to convert the index vox file in to a series of wave files and then use sox to convert the wave files to gsm files. Over all this works really well, the problem is that about 60 to 70 percent of the gsm files have some static or popping and clicking, on most of them it is in the silence at the end of the file. All that back story to ask this question: Are there any good utilities available for cleaning up gsm files? Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [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 ___ --Bandwidth 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] Voice mail server
Hi, how i have to do for receive a email with a alert from my voice mail? My doubt is what I put in serveremail in file voicemail.conf. I think is a email server, but can be see anyone? I searching one in the internet? Thanks and sory my english ___ --Bandwidth 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] MusiconHold
Hi, i configured Musiconhold and Works, but the sound is very low. I haved put the volume in the max, but is equal. I tested to my voice, and the sound is also low. exten=8000,1,Wait(2) exten=8000,2,Record(menu:gsm) exten=8000,3,Wait(2) exten=8000,4,Playback(menu) exten=8000,5,Hangup() when the musicaonhold is play e recieved this warning. exten = 6000,1,MusicOnHold() Executing MusicOnHold(SIP/2000-f7d9, pessoal) in new stack -- Started music on hold, class 'pessoal', on SIP/2000-f7d9 Feb 16 15:45:14 WARNING[8318]: interface.c:215 decodeMP3: Junk at the beginning of frame Please I need a suggestion, I NOT HAVE FXO, only two network card Thanks and sory my english ** My configuration: Extensions.conf exten = 6000,1,MusicOnHold() Zapata.conf musiconhold=default context=default musiconhold.conf [default] directory=/var/lib/asterisk/mohmp3/pessoal/ mode=files random= yes ___ --Bandwidth 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] Voice mail server
Carlos Jerónimo wrote: Hi, how i have to do for receive a email with a alert from my voice mail? You need a working installation of sendmail on your server. Then you append the email address of the users to the mailbox definitions in voicemail.conf like this: 1234 = 1234,Some User,[EMAIL PROTECTED] (See http://www.voip-info.org/wiki-Asterisk+config+voicemail.conf) My doubt is what I put in “serveremail” in file voicemail.conf. I think is a email server, but can be see anyone? I searching one in the internet? No, that's not a mail server. Just use the default serveremail=asterisk Regards, Philipp -- 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 Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --Bandwidth 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] New Community Blogs
Hello all, I'd like to introduce you to a new feature that we're opening up for all users on the AsteriskNOW.org website. Anyone with an account can now post a blog on the front page. This feature will give you the opportunity to post stories about what you've done with Asterisk and AsteriskNOW for everyone to see. After you log into AsteriskNOW.org, you will see a box on the right, containing a link that says create content. Click there, then blog entry on the main page. Fill out the boxes, and click submit, and then you're done. In the process of adding this feature, we have updated the Communications Rules located at http://www.asterisk.org/community/rules, so check those out before posting. Thanks, Aaron Daniel Community Relations Specialist [EMAIL PROTECTED] (256) 428-6010 ___ --Bandwidth 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] H extension don't work with parked calls
Hi all, I'm having a problem, with the h extension.
I have an application, when I call it check for the line requested and then
direct the call to a predefined context.
In this context I play a message (the message according to the line called)
and then park the call.
The dialplan does some other things, but my problem is that if I hung the
phone the h extension don't run, this is my dial plan
office]
include = check_voicemail
include = parking_lot
include = record_msgs
exten = fax,1,macro(RecibirFax)
exten = h,1,DeadAGI(end_logger.agi)
exten = s,1,answer()
;; pregunte por el caller id
exten = s,2,GotoIf($[${CALLERID(num)}]?4:3)
;; si no lo tiene entonces que lo cambie por 'Numero Privado'
exten = s,3,Set(CALLERID(all)=Numero Privado)
exten = s,n,SET(ARG1='2')
exten = s,n,AGI(logger.agi)
exten = s,n,hangup()
exten = ACC-4,1,playback(${SOUNDS}welcome-4)
exten = ACC-4,n,park(704)
exten = ACC-4,n,hangup
But the h extension is never called?
ideas?
--
==
Jonathan S.
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Re: [asterisk-users] SLA more than 100% ?
On 24/02/07, Tim Connolly [EMAIL PROTECTED] wrote: How does one answer more than 100% of the calls in less than 60 seconds? techsupp has 0 calls (max 20) in 'leastrecent' strategy (104s holdtime), W:0, C:3, A:2, SL:166.7% within 60s Probably talking out of my hat (I've never particularly looked at those figures), but might there have been some calls in progress at the start of the 60s which have finished? If it works on subtract for call started, add for call-stopped...? Andrew -- Linux supports the notion of a command line or a shell for the same reason that only children read books with only pictures in them. Language, be it English or something else, is the only tool flexible enough to accomplish a sufficiently broad range of tasks. -- Bill Garrett ___ --Bandwidth 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] Voice mail server
On Fri, Feb 23, 2007 at 11:33:30PM +0100, Philipp Kempgen wrote: Carlos Jerónimo wrote: Hi, how i have to do for receive a email with a alert from my voice mail? You need a working installation of sendmail A sendmail, actually. postfix, exim or whatever will also do. Or even nullmailer or a similar program with no local spool (though it is generally not recommended). -- 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] Asterisk callshops
Hello:
I have some questions regarding using a striped version of asterisk compiled
in a mips32 adsl router (probabilly broadcom 96348R Linux version 2.6.8.1
([EMAIL PROTECTED]) (gcc version 3.4.2)
I would like your comments on this dialplan
(do you think it will work)
[frombooths]
exten = _X.,1,DBQuery(blroute,(${CALLERID},${EXTEN}),(IS_BLOCKED|DESTINATION|
RATE|PERIOD))
exten = _X.,n,GotoIf(${IS_BLOCKED} = 1 ?blockedphone,s,1)
¿¿¿ exten = _X.,n,Answer ??? ; As per docs.. ¿Implications on ringtones???
and variable ANSWEREDTIME,
exten = _X.,n,SendText(${DESTINATION},${RATE})
exten = _X.,n,Dial(IAX2/user:[EMAIL PROTECTED]/${EXTEN},,M(counter,${RATE},
{PERIOD}))
exten = h,n,Hangup()
[macro-counter]
exten = s,1,Set(N=$[1])
exten = s,n,Set(PRICE=[${N}*${PERIOD}*{RATE}])
exten = s,n,Sendtext(${PRICE})
exten = s,n,While($[${DIALSTATUS} = ANSWER])
exten = s,n,While($[${ANSWEREDTIME} = $[${N}*${PERIOD}]])
exten = s,n,Set(N=$[${N}+1])
exten = s,n,Set(PRICE=$[${N}*${PRICE}])
exten = s,n,SendText(${PRICE})
exten = s,n,WhileEnd
exten = s,n,WhileEnd
[blockedphone]
¿¿¿ exten = s,1,Answer ???
exten = s,2,SendText(Telefono Bloqueado)
exten = s,3.Hangup (); probably some ringtone special or send some state
different that bussy if its posible
--
Francisco J. Pérez Botella
--
Francisco J. Perez Botella
tecnico-comercial tel: 669365228
647507437
email:[EMAIL PROTECTED]
Meridiam Phone
C/ Padre Mariana, 15 - 1º 03004 Alicante
Telf.: 965 201 550 / 902 947 884
Fax : 965 215 314 / 902 947 885
http://www.gruposati.com
--
Francisco J. Pérez Botella
___
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RE: [asterisk-users] Voice mail server
Thanks Philipp. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Philipp Kempgen Sent: sexta-feira, 23 de Fevereiro de 2007 22:34 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Voice mail server Carlos Jerónimo wrote: Hi, how i have to do for receive a email with a alert from my voice mail? You need a working installation of sendmail on your server. Then you append the email address of the users to the mailbox definitions in voicemail.conf like this: 1234 = 1234,Some User,[EMAIL PROTECTED] (See http://www.voip-info.org/wiki-Asterisk+config+voicemail.conf) My doubt is what I put in serveremail in file voicemail.conf. I think is a email server, but can be see anyone? I searching one in the internet? No, that's not a mail server. Just use the default serveremail=asterisk Regards, Philipp -- 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 Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --Bandwidth 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] Voice mail server
Thanks Philipp. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Philipp Kempgen Sent: sexta-feira, 23 de Fevereiro de 2007 22:34 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Voice mail server Carlos Jerónimo wrote: Hi, how i have to do for receive a email with a alert from my voice mail? You need a working installation of sendmail on your server. Then you append the email address of the users to the mailbox definitions in voicemail.conf like this: 1234 = 1234,Some User,[EMAIL PROTECTED] (See http://www.voip-info.org/wiki-Asterisk+config+voicemail.conf) My doubt is what I put in serveremail in file voicemail.conf. I think is a email server, but can be see anyone? I searching one in the internet? No, that's not a mail server. Just use the default serveremail=asterisk Regards, Philipp -- 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 Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --Bandwidth 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] b410p + fax (echo cancellation)
I really don't get it From several emails in this list archive, I had clearly understood that it is important to switch Echo Cancellation off for fax-channels, or faxing would not work properly. However, faxing (B410P ISDN bridged to TE410P PSTN) seems to work fine with EC at 256 taps on the B410P. I am confused; can anyone enlighten me? Thank you! Z. == Zoilo Gomez wrote: We have recently purchased a B410P Digium 4* ISDN-2 card with hardware EC. On the same server, I also have a regular Digium 4-channel PSTN-card (TDM410P ?), used to interface to some analog devices, a.o. 2 fax machines. For faxing, EC needs to be off (or so I understand from the archives). How can I switch EC off for an ISDN B-channel if a fax is coming in? Z. ___ --Bandwidth 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] b410p + fax (echo cancellation)
Zoilo Gomez wrote: I really don't get it From several emails in this list archive, I had clearly understood that it is important to switch Echo Cancellation off for fax-channels, or faxing would not work properly. However, faxing (B410P ISDN bridged to TE410P PSTN) seems to work fine with EC at 256 taps on the B410P. I am confused; can anyone enlighten me? Echo cancellation does not necessarily break faxing. However, depending upon how it is implemented it can. In general fax does not care about echo (as long as it is, indeed, more of a sidetone than an actual echo), and so it's generally good advice to tell people to disable echo cancellation on ATAs and other things when faxing is being used on them simply as a preventative measure. Lee. ___ --Bandwidth 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] ReceiveText()?
How do I receive text sent from SendText() application? Asterisk lists text capability, so SendText() is successful. But I don't see an application to actually use it. Yuan Liu ___ --Bandwidth 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] b410p + fax (echo cancellation)
On Friday 23 February 2007 8:35 pm, Zoilo Gomez wrote: However, faxing (B410P ISDN bridged to TE410P PSTN) seems to work fine with EC at 256 taps on the B410P. Generally speaking all modems (this includes POS machines and faxes) emit a tone which echo cancellers recognize and disable themselves for that call. -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] b410p + fax (echo cancellation)
Zoilo Gomez wrote: I really don't get it From several emails in this list archive, I had clearly understood that it is important to switch Echo Cancellation off for fax-channels, or faxing would not work properly. However, faxing (B410P ISDN bridged to TE410P PSTN) seems to work fine with EC at 256 taps on the B410P. I am confused; can anyone enlighten me? With EC on, modems might work, or they might not. If you have little or no echo, the EC actually does very little, so it won't significantly affect the one way at a time modems used for faxing. If you have a lot of echo, then having the EC switched on will degrade the modem signal. Things might still work, as a modem adapts itself to line conditions. You probably get more bit errors, but the fax will still get through. However, it doesn't take too much echo before the modems are pushed beyond the limits of what they can compensate for, and the receive side cannot extract the bit stream at all. So, the bottom line is it is not black and white, but EC off is more reliable. Regards, Steve ___ --Bandwidth 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 and Cisco PRI gateway config
Do you think it could have been done with another T1/E1Asterisk box between the Nortel PBX and the other Asterisk server ? Sorry, I do not understand exactly what you are asking. Do you mean using an Asterisk with PRI card instead of Cisco? If so, I have no experience with this. Which features would you then loose or gain, given current status of QSIG support in Asterisk ? In my case the Cisco did all the Q.sig work so Asterisk's Q.sig functionality was not used. __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
[asterisk-users] Any way to get rid of AEL created contexts?
show dialplan keeps showing contexts created by AEL. I tried deleting /etc/asterisk/extensions.ael but kept getting these messages in the Asterisk log: Feb 14 21:39:53 WARNING[6074] pbx_ael.c: Unable to open '/etc/asterisk/extensions.ael': No such file or directory Feb 14 21:39:53 WARNING[6074] pbx.c: Requested contexts didn't get merged Is there any way to delete or disable AEL? ___ --Bandwidth 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] Any way to get rid of AEL created contexts?
It would be something like noload = pbx_ael.so in /etc/asterisk/modules.conf later, PaulH On Sat, 2007-02-24 at 16:16 +1100, Eric Bishop wrote: show dialplan keeps showing contexts created by AEL. I tried deleting /etc/asterisk/extensions.ael but kept getting these messages in the Asterisk log: Feb 14 21:39:53 WARNING[6074] pbx_ael.c: Unable to open '/etc/asterisk/extensions.ael': No such file or directory Feb 14 21:39:53 WARNING[6074] pbx.c: Requested contexts didn't get merged Is there any way to delete or disable AEL? ___ --Bandwidth 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] Any way to get rid of AEL created contexts?
Eric Bishop wrote: show dialplan keeps showing contexts created by AEL. I tried deleting /etc/asterisk/extensions.ael but kept getting these messages in the Asterisk log: Feb 14 21:39:53 WARNING[6074] pbx_ael.c: Unable to open '/etc/asterisk/extensions.ael': No such file or directory Feb 14 21:39:53 WARNING[6074] pbx.c: Requested contexts didn't get merged Is there any way to delete or disable AEL? edit /etc/asterisk/modules.conf add noload = pbx_ael.so and stop/start asterisk ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Accessible documentation vor blind users
Hi Hi Is there any accessible ocumentation, ie plain text or html, how to configure Asterisk. The book 'Asterisk: The Future of Telephony'' is availablly only as and pdf document and is thus unreadable for a blind user. Any pointers welcome. You can still escape from the Gates of hell: Use Linux! -- arimo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
