[Asterisk-Users] Re: OT: SCALE 4x -- Call For Papers
Hello, The deadline for submissions to the SCALE 4x Call For Papers has been extended through Dec 3rd. If you are interested in presenting there is still time to get your proposals in. Regards, Ilan On 9/1/05, Ilan Rabinovitch [EMAIL PROTECTED] wrote: Hello, The call for papers for SCALE 4x, the 2006 Southern California Linux Expo, is now open. This event will be our fourth annual show. It will be held on Feb 11-12, 2006 at the Los Angeles Airport Westin. We are expecting 1,300+ in attendance this year. We are non-profit, community run Linux, open-source and free software conference. If you are working on something you believe the community would be interested in, please consider submitting a presentation as part of our call for papers. I am including details bellow. Past presentations are available online (including slides and audio in most cases): 2005 - http://www.socallinuxexpo.org/past/2005/hours.php 2003 - http://www.socallinuxexpo.org/past/2003/presentations.php 2002 - http://www.socallinuxexpo.org/past/2002/presentations.php If you have any questions please feel free to call the Call For Papers team at cfp @ socallinuxexp.org CFP Link: http://www.socallinuxexpo.org/pr/pr_20050620.php CFP PDF: http://www.socallinuxexpo.org/pr/cfp4x.pdf Best regards, Ilan Rabinovitch Conference Chair Southern California Linux Expo http://www.socallinuxexpo.org 2006 Southern CAlifornia Linux Expo The USC, Simi/Conejo, and UCLA Linux User Groups are proud to announce the 4th annual Southern California Linux Expo scheduled for February 11-12, 2006 at the Westin Hotel near the Los Angeles International Airport. Building on the tremendous success of last three years' SCALE, we will continue to promote Linux and the Open Source Software community. We invite you to share your work on Linux and Open Source projects with the rest of the community as well as exchange ideas with some of the leading experts in these field. Details about SCALE 4X as well as archives for the last three years can be found at http://www.socallinuxexpo.com. Topics of interest include, but are not limited to: * Linux kernel * Linux Networking * Linux for embedded systems * Linux for Desktops * LAMP * Multimedia in Linux * Security in Linux * VoIP * Wireless tools in Linux * Linux Games * GIMP other graphics software * Administration techniques for specific distributions * Custom Configurations * Linux Deployments and experiences: Case studies * Open source Licensing * Government policies with Open Source * Other open source projects The proposals should comprise a 1-page (maximum) description containing the following: 1] Title for the talk. 2] Name, Affiliation, Bio, a passport size picture (optional) and contact email address of the Presenter. 3] What will be covered? A bulleted list of the main points of the presentation will be ideal. Please include enough detail as will be necessary. 4] Any specific requirements needed for the presentation other than an overhead projector and a microphone. Presentations are alloted a time slot of about 45 minutes. All proposals are to be sent to [EMAIL PROTECTED] Important Dates: 20 Jun, 2005: CFP Opens 20 Nov, 2005: Last date for abstracts/proposals 20 Dec, 2005: Last date for notification of acceptance 11 Feb, 2006: Conference starts ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Test numbers for ENUM (e164.arpa, e164.org, etc.)
www.enum-test.at (english translation is comming soon) klaus John Todd wrote: I'm looking to build a decent list of test numbers which have ENUM resolution. The numbers I'm looking for should go to a recording, an echo test, or some other feature which does NOT lead to a human. These will be for manual or semi-automatic testing (i.e.: we'll test 10 times in a day, but we won't test continuously.) Any public ENUM-ish tree is fine, but I'm really shooting for e164.arpa. I'm especially interested in: - numbers in +87810 in e164.arpa (does anyone actually USE this space, considering the unpleasant commercialism of the project now?) - numbers in +1 in e164.arpa (test only, of course) - any test numbers at all, anywhere that resolve within the DNS - I'm still interested in .de and .at numbers, but I actually have several of these now I will make this list public in the http://www.voip-info.org/wiki/view/Phone+Numbers page unless otherwise noted in the message. The purpose of this test is to see what DNS resolution times look like with valid numbers, and then add to that post-dial-delay in real-world situations to get an idea of total setup time. JT ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sip URL peering
There is a new ietf WG to come which deals with peering issues. It's called SPEER (formerly VOIPEER) The list archive is at http://darkwing.uoregon.edu/~llynch/voipeer/ minutes from last ietf meeting: http://www3.ietf.org/proceedings/05nov/minutes/voipeer.html regards klaus Chris Hills wrote: Wolfgang S. Rupprecht wrote: One thing I haven't seen get much airtime on the digium lists is sip URL-based peering. I imagine many of us have far more asterisk extensions than PSTN numbers. It would be really nice to be able to do something like call [EMAIL PROTECTED] from [EMAIL PROTECTED] It looks like all or most of the pieces are in place, but I don't see folks discussing it much. Is no-one else interested in this? Perhaps you would be interested in TRIP (telephony routing over ip)? Each organisation can apply for an ITAD number, just like a domain. TRIP numbers take the form extension*itad, for example, 1234*222. As you can no doubt surmise, TRIP numbers can be dialled from a regular telephone handset. For more information, please see the following documents:- http://www.iana.org/assignments/trip-parameters http://www.ietf.org/rfc/rfc3219.txt Regards -- Chris Hills IT Services North East Worcestershire College ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk server behind NAT, and SIP clinet behind another NAT.
Asterisk server behind NAT,and SIP clinet behind another NAT. SIP.conf have set NAT=yes, SIP client can register with Asterisk server, but can not hearing anything.. PLS help me, how to resolve this trouble,, As refer to the item 9 http://www.voip-info.org/wiki-Asterisk+SIP+NAT+solutions I can not register with Asterisk server too, how this happen.. _ Dont just search. Find. Check out the new MSN Search! http://search.msn.click-url.com/go/onm00200636ave/direct/01/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: [Serusers] Re: open letter
Doug, You have ever post this mail. Harry Others have tried to explain it too you, but I don't think you fully understand. Maybe it is a language issue. Your follow-up posts come across as demanding. When I read your posts, I feel like you are criticizing people for not having responded to you. It is like you feel they have done something wrong. This probably isn't what you mean, but that is how it seems. ___ Serusers mailing list [EMAIL PROTECTED] http://mail.iptel.org/mailman/listinfo/serusers ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] how to configure analog phone
Download Asterisk cvs-head cd /usr/src export CVSROOT=:pserver:[EMAIL PROTECTED]:/usr/cvsroot cvs login - the password is anoncvs. cvs checkout zaptel asterisk MAKE ZAPTEL cd /usr/src/zaptel make clean make linux26 make install and if you want asterisk to start automatically add make config MAKE ASTERISK cd /usr/src/asterisk make clean make install make samples and if you want asterisk to start automatically add make config SIMPLE CONFIG /etc/zaptel.conf loadzone = us fxoks=1 /etc/asterisk/zapata.conf [channels] signalling=fxo_ks language=en context=incoming channel = 1 /etc/asterisk/extensions.conf [general] [incoming] exten = _X.,1,Answer exten = _X.,2,Playback(invalid) exten = _X.,3,Hangup STARTING ASTERISK modprobe zaptel modprobe wcfxo ztcfg asterisk -vvvcg TESTING Pickup your handset connected to the FXS port, you should hear a dialtone, then dial . You should hear I am sorry that is not a valid extension, please try again __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] open letter (2)
Advice me and I'll stop to mail my question. LOL ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] open letter (2)
Dear users, This letter is addressed to the most experienced users for the ser openser and asterisk projects. Advice me and I'll stop to mail my question. How a session between two user agents behind nat could stay in the path ? Harry Kinds Regards |register || register | agent1 asterisk| |ser/nat box || | 200 OK ||200 OK | agent2 One box --- | | | | asterisk pbx | | | | ||| | | ---- | | SER ||NAT box | private network | ---- --- ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Re: [Asterisk-Users] Multiple HFC-PCI cards in mixed modes (TE+NT) won't work!...
Maybe it's worth a try, using chan_mISDN (experimental, but works!).. You can find the how-to pdf (for beronet, hfc, etc.. cards) on http://www.beronet.com/downloads/. There also is an install-script that helps you through the installation, I have gotten it to work with a junghanns card and 1x HFC pci card. Didn't have a 2nd hfc around to try back then.. I you have results (good/bad) keep the list (or me) posted :) The mISDN and chan_misdn supports multiple HFC-PCI cards in mixed modes. We are using this drivers in our distribution ( http://amatisoft.homelinux.com/amatix.html ). -- Amatisoft SRL http://amatisoft.homelinux.com __ Yahoo! FareChase: Search multiple travel sites in one click. http://farechase.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk and DrayTek Vigor2600VGi
Hi all, I'm trying to configure a remote user with a DrayTek 2600Vgi. The setup looks like this. [SIPphone]--[Asterisk]--[Firewall]---[VPN]---[DrayTek]--[Analog-phone] I can place calls to the DrayTek and recieve calls from the analog phone. However, the calling party does not hear the called party (one way audio). The audio for the remote user works fine. VPN works fine too and i have no drops in my fw-logs. my sip.conf looks like this. (ext 2005 is my DrayTek and ext 2006 is the local sip user) [general] context=default port=5060 bindaddr=0.0.0.0 srvlookup=yes [2006] type=friend callerid=OptiPoint600 2006 context=international host=dynamic user=2006 secret=x dtmfmode=rfc2833 disallow=all allow=alaw allow=ulaw [2005] type=friend canreinvite=no host=dynamic user=2005 secret=x dtmfmode=rfc2833 callerid=Draytek 2005 context=international disallow=all allow=alaw allow=ulaw i try to play around with externip, localnet, nat and canreinvite but still have the same issue. Sporadically i can see a Maximum retries exceeded on call ... for seqno 102 (Non-critical Request on the CLI. I have also replaced the analog phone with a softclient on my laptop connected behind the vigor, but have the same problem. Has anybody managed to get a similar setup running? any ideas, suggestions are wellcome... thx in advance... richard. __ Yahoo! FareChase: Search multiple travel sites in one click. http://farechase.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ver1.2 installation problem
I had this problem with Fedora. I updated the kernel to the latest one available for core 3 and changes the links to point to the new source code. It worked fine then. Regards Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bartosz Wegrzyn - asterisk Sent: 23 November 2005 03:42 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] ver1.2 installation problem Hi, After I compile asterisk v.1.2 is tells me that last thing to do is to make install. Unfortunately it goes it to loop after I type make install this is the loop: else \ mv include/asterisk/version.h.tmp include/asterisk/version.h ; \ fi rm -f include/asterisk/version.h.tmp build_tools/mkdep -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g3 -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -fomit-frame-pointer acl.c aescrypt.c aeskey.c aestab.c alaw.c app.c asterisk.c ast_expr2.c ast_expr2f.c astmm.c autoservice.c buildinfo.c callerid.c cdr.c channel.c chanvars.c cli.c config.c cryptostub.c db.c devicestate.c dlfcn.c dns.c dnsmgr.c dsp.c enum.c file.c frame.c fskmodem.c image.c indications.c io.c jitterbuf.c loader.c logger.c manager.c md5.c muted.c netsock.c pbx.c plc.c poll.c privacy.c rtp.c say.c sched.c slinfactory.c srv.c strcompat.c tdd.c term.c translate.c ulaw.c utils.c build_tools/make_version_h include/asterisk/version.h.tmp if cmp -s include/asterisk/version.h.tmp include/asterisk/version.h ; then echo; else \ mv include/asterisk/version.h.tmp include/asterisk/version.h ; \ fi rm -f include/asterisk/version.h.tmp build_tools/mkdep -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g3 -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -fomit-frame-pointer acl.c aescrypt.c aeskey.c aestab.c alaw.c app.c asterisk.c ast_expr2.c ast_expr2f.c astmm.c autoservice.c buildinfo.c callerid.c cdr.c channel.c chanvars.c cli.c config.c cryptostub.c db.c devicestate.c dlfcn.c dns.c dnsmgr.c dsp.c enum.c file.c frame.c fskmodem.c image.c indications.c io.c jitterbuf.c loader.c logger.c manager.c md5.c muted.c netsock.c pbx.c plc.c poll.c privacy.c rtp.c say.c sched.c slinfactory.c srv.c strcompat.c tdd.c term.c translate.c ulaw.c utils.c build_tools/make_version_h include/asterisk/version.h.tmp if cmp -s include/asterisk/version.h.tmp include/asterisk/version.h ; then echo; else \ mv include/asterisk/version.h.tmp include/asterisk/version.h ; \ fi rm -f include/asterisk/version.h.tmp build_tools/mkdep -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g3 -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -fomit-frame-pointer acl.c aescrypt.c aeskey.c aestab.c alaw.c app.c asterisk.c ast_expr2.c ast_expr2f.c astmm.c autoservice.c buildinfo.c callerid.c cdr.c channel.c chanvars.c cli.c config.c cryptostub.c db.c devicestate.c dlfcn.c dns.c dnsmgr.c dsp.c enum.c file.c frame.c fskmodem.c image.c indications.c io.c jitterbuf.c loader.c logger.c manager.c md5.c muted.c netsock.c pbx.c plc.c poll.c privacy.c rtp.c say.c sched.c slinfactory.c srv.c strcompat.c tdd.c term.c translate.c ulaw.c utils.c build_tools/make_version_h include/asterisk/version.h.tmp if cmp -s include/asterisk/version.h.tmp include/asterisk/version.h ; then echo; else \ mv include/asterisk/version.h.tmp include/asterisk/version.h ; \ fi rm -f include/asterisk/version.h.tmp build_tools/mkdep -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g3 -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -fomit-frame-pointer acl.c aescrypt.c aeskey.c aestab.c alaw.c app.c asterisk.c ast_expr2.c ast_expr2f.c astmm.c autoservice.c buildinfo.c callerid.c cdr.c channel.c chanvars.c cli.c config.c cryptostub.c db.c devicestate.c dlfcn.c dns.c dnsmgr.c dsp.c enum.c file.c frame.c fskmodem.c image.c indications.c io.c jitterbuf.c loader.c logger.c manager.c md5.c muted.c netsock.c pbx.c plc.c poll.c privacy.c rtp.c say.c sched.c slinfactory.c srv.c strcompat.c tdd.c term.c translate.c ulaw.c utils.c build_tools/make_version_h include/asterisk/version.h.tmp if cmp -s include/asterisk/version.h.tmp include/asterisk/version.h ; then echo; else \ mv include/asterisk/version.h.tmp include/asterisk/version.h ; \ fi rm -f include/asterisk/version.h.tmp build_tools/mkdep -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g3 -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -fomit-frame-pointer acl.c aescrypt.c aeskey.c aestab.c alaw.c app.c asterisk.c ast_expr2.c ast_expr2f.c astmm.c autoservice.c buildinfo.c callerid.c cdr.c channel.c chanvars.c cli.c config.c cryptostub.c db.c devicestate.c dlfcn.c dns.c dnsmgr.c dsp.c
Re: [Asterisk-Users] Multiple HFC-PCI cards in mixed modes (TE+NT) won't work!...
On Wed, November 23, 2005 7:21, Tzafrir Cohen said: On Tue, Nov 22, 2005 at 09:20:23PM +0100, Francesco Peeters wrote: I have seen several claims that it can be done: Multiple HFC-PCI cards running both BRI_CPE_PTMP and BRI_NET_PTMP simultaneously, but *nobody* has been able yet to tell me the exact setup to make it work... My asterisk server breaks as soon as I turn one of the two cards in to an NT card, which I suspect is an issue in the (Florz-patched) BriStuffed chan_zap.so module, but as my cards aren't original Junghanns', they obviously aren't supported by Junghanns... (ZapHFC btw shows both cards active, one as TE Master, the other as NT slave, so that seems to be OK! I therefore doubt it is the Florz patch!) And you're very scarse on details here. How about your zaptel.conf and the relevant prts of your zapata.conf? I had those in previous posts to which nobody responded... I'll repeat below! Also: if you want just one of the cards in NT mode and the other in TE mode, how can you be sure which one os the first and which is the second? 1) Put in both cards 2) Activate only the first in TE mode 3) Test which one responds with D-channel up/down messages when connected to the PSTN 4) The other one is the second one, which will be running in NT mode... ;-) -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is zaptel.conf # hfc-s pci a span definition # most of the values should be bogus because we are not really zaptel loadzone=nl defaultzone=nl span=1,1,3,ccs,ami bchan=1-2 dchan=3 span=2,1,3,ccs,ami bchan=4-5 dchan=6 end of zaptel.conf zapata.conf ; ; Zapata telephony interface ; ; Configuration file [channels] ; ; Default language ; language=nl ; ; Default context ; ; switchtype = euroisdn rxwink=300 usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 nationalprefix = 0 internationalprefix = 00 faxdetect=incoming group=1 callgroup=1 pickupgroup=1 ; p2mp TE mode ;signalling = bri_cpe_ptmp ; p2p TE mode ;signalling = bri_cpe ; p2mp NT mode ;signalling = bri_net_ptmp ; p2p NT mode ;signalling = bri_net pridialplan = dynamic prilocaldialplan = local nationalprefix = 0 internationalprefix = 00 echocancel=yes echotraining = 100 echocancelwhenbridged=yes signalling = bri_cpe_ptmp immediate=no overlapdial=yes group = 1,2,3,4 callgroup=1,2,3,4 pickupgroup=1,2,3,4 context=from-pstn channel = 1-2 signalling = bri_net_ptmp ;signalling = pri_net_ptmp immediate=no overlapdial=yes group = 11,12,13,14 callgroup=11,12,13,14 pickupgroup=11,12,13,14 context=ext-local channel = 4-5 ;Include genzaptelconf configs #include zapata-auto.conf ;Include AMP configs #include zapata_additional.conf end of zapata.conf -- Francesco Peeters GPG Key = AA69 E7C6 1D8A F148 160C D5C4 9943 6E38 D5E3 7704 If your program doesn't recognize my signature, please visit http://www.CAcert.org/index.php?id=3 to retrieve the Root CA certificate. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: [Users] open letter (2)
Hi Klaus, Please do not cross post. Split your problems into smaller problems and ask them on the correspondig list. I mail my question to asterisk, openser ser lists After all your emails, I still have no glue what your scenario is. Why do you want to host ser+asterisk+NAT on the same device? pass through I agree my english is not very good sorry i try my best . Asterisk don't provide IM/presence unlike ser however ser don't provide telephony features like MOH ACD call parked IVR and more I want my sip agents to provide these features. Ser handle sip routing asterisk telephony features . Should the Asterisk/ser be reachable also from the public interface? If not, why do you need NAT traversal at all? In fact i have got a single machine for my tests . Ser handle sip routing so incoming or outgoing requests pass through SER not directly to asterisk . I need nat support for sip agents behind nat. Why do you use both? Asterisk can also do NAT traversal. For how many users is the setup? I think asterisk support 255 users klaus harry gaillac wrote: Dear users, This letter is addressed to the most experienced users for the ser openser and asterisk projects. Advice me and I'll stop to mail my question. How a session between two user agents behind nat could stay in the path ? Harry Kinds Regards |register || register | agent1 asterisk| |ser/nat box || | 200 OK ||200 OK | agent2 One box --- | | | | asterisk pbx | | | | ||| | | ---- | | SER ||NAT box | private network | ---- --- ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ Users mailing list Users@openser.org http://openser.org/cgi-bin/mailman/listinfo/users ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Wildcard FXO takes too long to answer incoming calls
Dear Adam,
Thank you very much for reply! l try it and let you know. By the way, this is my incoming context in my extensions.conf:
[from-pstn-reghours-nofax]
exten = s,1,SetVar(intype=${INCOMING})
exten = s,2,Cut(intype=intype,-,1)
exten = s,3,GotoIf($[${intype} = EXT]?4:5)
; If INCOMING starts with EXT, then assume its an
extension
exten = s,4,Goto(ext-local,${INCOMING:4},1)
exten = s,5,GotoIf($[${intype} = GRP]?6:7) ; If INCOMING starts with GRP, then assume its a ring group
exten = s,6,Goto(ext-group,${INCOMING:4},1)
exten = s,7,GotoIf($[${intype} = QUE]?8:11) ;queue
exten = s,8,Answer
; answer call
before queue
exten = s,9,Wait(1)
exten = s,10,Goto(ext-queues,${INCOMING:4},1)
exten = s,11,Answer
; answer call
before auto attendant
exten = s,12,Wait(1)
exten = s,13,Goto(${INCOMING},s,1)
; not EXT or
GR1 - it's an auto attendant
exten = fax,1,Goto(ext-fax,in_fax,1)
exten = h,1,Hangup
Again, thank you for your help guys!
kenjieOn 11/21/05, Adam Goryachev [EMAIL PROTECTED] wrote:
On Sat, 2005-11-19 at 18:02 +0100, Michael Kenjie Nukui wrote: Hi, i have this Wildacard FXO in my [EMAIL PROTECTED] box, connected to POTS.When i make an incoming call, it takes about 8 to 10 rings before my card pick up the incoming call and answers it.Here is my
config.Can somebody help me? Any assistance will be highly appreciated. Thank you. zapata.conf usecallerid=yesSet this to nousecallerid=noAlso show us your
extensions.conf for the incoming context.Regards,Adam___--Bandwidth and Colocation sponsored by Easynews.com --
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Re: [Asterisk-Users] Asterisk and DrayTek Vigor2600VGi
Hello richard, Wednesday, November 23, 2005, 10:46:03 AM, you wrote: rC Hi all, rC I'm trying to configure a remote user with a DrayTek rC 2600Vgi. The setup looks like this. rC [SIPphone]--[Asterisk]--[Firewall]---[VPN]---[DrayTek]--[Analog-phone] I tried a similar setup some times ago and it was working, have you put the private ip address of the asterisk box in the vigor setup ? Can you ping the private address of the vigor from the asterisk box and viceversa ? Hope it helps ! -- Best regards, Alessiomailto:[EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Aastra 1.3 firmware
Title: Aastra 1.3 firmware Has anyone had any luck with the BLF option yet? I have set up as per the manual/front end, configured the hints in Asterisk and nothing shows. Regards Lee ###This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange.For more information, connect to http://www.f-secure.com/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] queue problem
hi guys. Im having a problem with queue setup. btw Im using Asterisk CVS-v1-0-11/21/05-20:21:21 to be exact. and im using AMP-1.10.010. I selected ringall in ring strategy. but when the calls come it it like random ringing it will ring to the 1st phone after 2 rings it will ring 2nd and 3rd then 3rd and 4th. can i implement something that if a call get it 4 phones will ring simultaneously until someone picked up? -- Regards,Mark Quitoriano, CCNAFan the flame...http://www.spreadfirefox.com/?q=user/registerr=19441 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and DrayTek Vigor2600VGi
Hi Alessio [SIPphone]--[Asterisk]--[Firewall]---[VPN]---[DrayTek]--[Analog-phone] I tried a similar setup some times ago and it was working, have you put the private ip address of the asterisk box in the vigor setup ? Can you ping the private address of the vigor from the asterisk box and viceversa ? I am able to ping the private addr of the vigor from * and of couse viceversa. The vigor setup seems to be ok (vpn is up and *sip show peers* shows that the vigor is registred.). I can also call from and to Asterisk, so the signalisation is ok. I have only problem with RTP packets (one way audio) anyhow thx for the feedback... __ Yahoo! FareChase: Search multiple travel sites in one click. http://farechase.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Aastra 1.3 firmware
Title: Aastra 1.3 firmware As always right after asking it works Lee From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lee ArcherSent: 23 November 2005 11:09To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] Aastra 1.3 firmware Has anyone had any luck with the BLF option yet? I have set up as per the manual/front end, configured the hints in Asterisk and nothing shows. Regards Lee ###This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange.For more information, connect to http://www.f-secure.com/ ###This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange.For more information, connect to http://www.f-secure.com/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dial ZAP with group (g2) erroneously says call answered when it is still ringing
Oh boy :(. As Roman politely explained in a private email... I was using ports 1 and 2 thinking they were the outbound fxs ports :(. That's it, these glasses are going, and no more testing from home :). When I switched to testing with ports 3 and 4, everything worked the same as G2. Not of course as cute as what I had hoped for when I see the local telco can do something like Dial(ZAP/g2/SIP/[EMAIL PROTECTED]) and have it wait 'til the correct phone is answered :(. Thanks to C F for the c option but my goal was to just have the 4 digit number call folks with and without SIP. I would not expect users to know to press #. I don't think dvlinedetect will quite cut it either. callprogress looked promising, but, alas, as many others have found, it hangs up after timeout seconds. I'll keep digging :). Thanks again everyone, JES James B. MacLean wrote: Hi C F, I am not well versed in this level of telephony or Asterisk, so please bare with me :). My setup is really typical. Bought the digium card with 4 ports. 2 fxs / 2 fxo. The 2 fxo's are connected directly to phones, belong to group 1 according to zapata.conf, and exist as fxoks=1-2 in /etc/zaptel.conf. The 2 fxs ports are connected to the telco, belong to group 2 according to zapata.conf, and are setup as fxsks=3-4 in zaptel.conf. Dial(Zap/1/SIP/[EMAIL PROTECTED],15,r) works as expected, Dial(Zap/2/SIP/[EMAIL PROTECTED],15,r) works as expected but: Dial(Zap/g2/SIP/[EMAIL PROTECTED],15,r) Rings once and reports answered to Asterisk. Does this support what you are explaining? I'm honestly confused by how an fxs module operates as an fxo module? Thanks for any more direction you might have, JES begin:vcard fn:James B MacLean n:MacLean;James B org:Education;ITS Technical Services adr:;;;Halifax;NS;;Canada email;internet:[EMAIL PROTECTED] url:http://www.ednet.ns.ca/~macleajb version:2.1 end:vcard smime.p7s Description: S/MIME Cryptographic Signature ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Multiple HFC-PCI cards in mixed modes (TE+NT) won't work!...
On Tue, 2005-11-22 at 22:04 +0100, Kristof Hardy wrote: Maybe it's worth a try, using chan_mISDN (experimental, but works!).. You can find the how-to pdf (for beronet, hfc, etc.. cards) on http://www.beronet.com/downloads/. There also is an install-script that helps you through the installation, I have gotten it to work with a junghanns card and 1x HFC pci card. Didn't have a 2nd hfc around to try back then.. I you have results (good/bad) keep the list (or me) posted :) I'm using chan_misdn successfully with two HFC cards; one in NT mode and one in TE mode. -- dwmw2 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re[2]: [Asterisk-Users] Asterisk and DrayTek Vigor2600VGi
Hello richard, Wednesday, November 23, 2005, 12:34:33 PM, you wrote: rC Hi Alessio rC [SIPphone]--[Asterisk]--[Firewall]---[VPN]---[DrayTek]--[Analog-phone] I tried a similar setup some times ago and it was working, have you put the private ip address of the asterisk box in the vigor setup ? Can you ping the private address of the vigor from the asterisk box and viceversa ? rC I am able to ping the private addr of the vigor from * rC and of couse viceversa. The vigor setup seems to be ok rC (vpn is up and *sip show peers* shows that the vigor rC is registred.). I can also call from and to Asterisk, rC so the signalisation is ok. I have only problem with rC RTP packets (one way audio) I'm having 10 peers over vpn vith 10 vigor in a customer setup, here is a sample of my sip.conf [general] port = 5060 ; Port to bind to bindaddr = 192.168.0.3 ; Address to bind SIP channel to context = default ; Default context for incoming calls ;srvlookup = yes; Enable DNS SRV lookups on outbound calls ;pedantic = yes ; Enable slow, pedantic checking for Pingtel ;tos=lowdelay ; IP QoS parameter, either keyword or value disallow=all ;allow=all ;allow=gsm allow=g729 ;allow=ilbc ;allow=ulaw ;allow=alaw [3200] ;LA SPEZIA 1 ;DRAYTEK VIGOR 2600 type=friend host=dynamic username=3200 secret=* canreinvite=yes context=sip qualify=yes I will also suggest to nail up the vpn connection from the vigor and upgrading the vigor firmware . Wish you luck! rC __ rC Yahoo! FareChase: Search multiple travel sites in one click. rC http://farechase.yahoo.com rC ___ rC --Bandwidth and Colocation sponsored by Easynews.com -- rC Asterisk-Users mailing list rC Asterisk-Users@lists.digium.com rC http://lists.digium.com/mailman/listinfo/asterisk-users rC To UNSUBSCRIBE or update options visit: rChttp://lists.digium.com/mailman/listinfo/asterisk-users -- Best regards, Alessiomailto:[EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] presence and Asterisk crash
Hi all. Ive got Asterisk CVS Head running on Fedora Core 3. It has been running for 4 months with no particular problem. Recently I tried to enable presence. On dialplan I added hint extensions for all my SIP users and on my Eyebeam clients (v. 1.1 3008q) I set Peer-to-Peer presence mode. Presence works right, but when an incoming or outogoing call is answered, Asterisk crashes with the following message: Ouch ... error while writing audio data: : Broken pipe Segmentation fault I tried to restart Asterisk many times but it always stop with this message. As I disable presence (on Eyebeam clients, not even in Asterisk dial plan) Asterisk stays on. Is this a bug or do I miss something with presence? Thank you, _fangi_ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Detect alternate line in Broadvoice inbound context
Just as a followup, you need to be a bit careful and test this out if
you make any changes to your Broadvoice account. I added a second
virtual number and they switched the number that was previously
specifying a distinctive ring of Bellcore-dr3 to Bellcore-dr4. The last
number added now specifies Bellcore-dr3. Sending the number would be so
much more reliable...
MARK.
Mark Hulber wrote:
Ok, your solution does work but in looking at my console output I saw
that SIPGetHeader was deprecated for the new dialplan function
SIP_HEADER. Below is my modification. You don't need a priority+101.
exten = 212999,1,Set(Var_Alert=${SIP_HEADER(Alert-Info)})
exten = 212999,n,GotoIf($[${Var_Alert} =
http://127.0.0.1/Bellcore-dr3]?x-916999,1:x-212999,1)
In this case, the 212 number is the primary number.
Thanks,
MARK.
Samy Antoun wrote:
Mark,
1. Make sure that SIPGetHeader application is registered
CLI show application SIPGetHeader
if it is registered you'll get
-= Info about application 'SIPGetHeader' =-
[Synopsis]
Get a SIP header from an incoming call
[Description]
SIPGetHeader(var=headername):
Sets a channel variable to the content of a SIP header
Skips to priority+101 if header does not exist
Otherwise returns 0
If not,
Your application(s) is (are) not registered
If the application is not registered, I can't recommend anything for
you, I had an Asterisk system with ver 1.0 (no SIPGetHeader) and I
tried to patch it with any of the following with no luck:
http://bugs.digium.com/bug_view_page.php?bug_id=0002838
http://bugs.digium.com/view.php?id=2924
If you have it registered, here is a sample of my setup:
[bvdr]
exten = s,1,Answer exten = s,2,DigitTimeout(5) exten =
s,3,ResponseTimeout(10) exten =
s,4,SIPGetHeader(Var_Alert=Alert-Info) exten =
s,5,GotoIf($[${Var_Alert} =
http://127.0.0.1/Bellcore-dr3]?ext-local,320,1) exten =
s,6,Goto(ext-local,200,1)
This setup for ONE Distinctive Ring only (Bellcore-dr3), if you have
more than one, you can use sip debug to retrieve the header
information
The BEST reference for this subject is:
http://voxilla.com/PNphpBB2-viewtopic-t-3935-highlight-dring1.html
Hope this helps
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[Asterisk-Users] Calling lines
hi guys, Isit possbile to show busy lines from tthe asterisk to be shown on cisco phones at the function buttons? I have cisco 7970 (snom phones have the same) and i want to have some numbers at the keys and if this number ist talking i want to see that. Normal like isdn pbx in normal way with system-telephones. regards rene ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 7960 audio quality when calling remote asterisk box
Hello all, I've been doing some testing with the 7960s I have here calling into a remote asterisk box (1.0.9). Audio quality on the 7960 is perfect when I call to other extensions on my local asterisk (1.2.0), but when I place calls to users on the remote box (boxes are linked via IAX2) audio quality drops massively - the party at the other end can hear what I'm saying perfectly, but I can barely make out one word in three. I then tried the same thing using a sip phone, and the audio problems aren't there at all. To summarize: audio problems: 7960 - local asterisk (1.2) - remote asterisk (1.0.9) - sip phone no problems: sip phone - local asterisk (1.2) - remote asterisk (1.0.9) - sip phone no problems: 7960 - local asterisk (1.2) - sip phone no problems: 7960 - local asterisk (1.2) - pstn I've tried disabling the IAX2 jitter buffer on both asterisks and forcing both of them to use the same codec, all without success. I'd be grateful for any hints as to which options I should check. Thanks in advance. Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited This email is made from 100% recycled electrons ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ISDN cards using CAPI interface
Hi all, if you configure a passive ISDN card with CAPI support, does it mean that can be used with asterisk (using chan-capi, of course) ?? Thanks ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Clearwire and Asterisk
Clearwire block the voip port 2005/11/23, Justin Newman [EMAIL PROTECTED]: Has anyone had problems using Clearwire, VOIP, and/or Asterisk? Just curious... ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] open letter (2)
On Wed, 2005-11-23 at 10:34 +0100, harry gaillac wrote: Advice me and I'll stop to mail my question. That almost sounds like a threat. Do you really think you motivate people to answer you this way? Since you asked this question already so many times perhaps it's time to hire a paid consultant. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: [Users] open letter (2)
--- Klaus Darilion [EMAIL PROTECTED] a écrit : Hi Harry! As this emails are on-topic you should cc: to the list. harry gaillac wrote: In fact the problem is in contact sip header field (private ip) agent send ReGISTER to SER (outbound proxy) which one send REGISTER to ASTERISK . Asterisk register agent with AOR sip:[EMAIL PROTECTED] ip When agent send INVITE to an other agent ASTERISK use AOR sip:[EMAIL PROTECTED] ip but the firewall don't allow this Asterisk SHOULD resend INVITE to SER. Does SER is able to rewrite contact field in SIP HF? Which IPaddress:port do you want to have in the REGISTER's Contact: header sent from ser to Asterisk? in fact i wish to replace all private ip in the contact field with the public ip of ASTERISK Harry klaus Regards Thanks for your advices Harry --- Klaus Darilion [EMAIL PROTECTED] a écrit : harry gaillac wrote: Have you ever used SIP clients with presence and IM? I suggest to setup ser (without Asterisk) just to test the IM features. SIP based IM/presence implementations are very poor yet. I've done it And what were your experiences? Which clients do you use? Polycom IP300 In your picture, the NAT router is on the same PC as ser and asterisk. Is this correct? this is correct It would be a good idea to split things. This is a rather complicated setup. what scenario do you have? Are all the users behding the same NAT (in the same subnet) and you provide VoIP within this network (e.g. an enterprise) or do you have external users (e.g. like iptel or freeworlddialup)? in fact both asterisk+ser private net=nathelper ==nat===private net nat box || internet== I suggest: 1. Asterisk, ser and the RTP proxy 8rtpproxy or mediaproxy) should listen only on the public interface (this really must be a routable public IP address, no private). SER asterisk listen on public ip 2. Setup the firewall (e.g. iptables) correctly to allow traffic from/to ser, asterisk and the RTP proxy Done 3. setup ser according the getting started document on onsip.org. AFAIK this document contains hints how to route to a gateway. Reuse this part of the config to route certain calls to the asterisk box. Done 4. Try to solve things step by step: - REGISTER should work fine from Internet and LAN - Calls from Internet clients to Internet clients - Calls from LAN clients to LAN clients - Calls from LAN clients to Internet clients (and vice versa) - now try to add asterisk, e.g. calling a certain number will be routed to asterisk and starts the echo application If all the above works (DO NOT start integrating the asterisk as long as basic SIP call do not work!), you can implement your setup. 5. Do really read every word in the getting started document, if things are unclear read it again. 6. Do not post how to make this setup. Ask small questions addressing particular (small) problems. 7. Post to the related list. - do not post to developer lists - if you use ser, post to ser's list - if you use openser, post to openser's list - if you have an asterisk problem, ask at the asterisk list (e.g. you want to solve NAT traversal and registration with ser. Thus, do not ask this kind of questions at the asterisk list). 8. always remember that this support is voluntary 9. If you don't find the proper english word, look into the dictionary instead of using another word which might also have other meanings. 10. Go and buy an english SIP book. (this will you help to learn the english terms for all the SIP stuff) 11. use ngrep to watch the SIP call flow # ngrep -t -d any port 5060 regards klaus === message truncated === ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] open letter (2)
harry gaillac wrote: Dear users, This letter is addressed to the most experienced users for the ser openser and asterisk projects. Advice me and I'll stop to mail my question. Is that something like, Tell me what I want to know and I'll go away? Doug ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] installing Asterisk from source
Daniel Mikusa wrote: Look in the Makefile for the variables 'INSTALL_PREFIX' and 'PREFIX' they control where Asterisk is installed. Dan Jeremy Jones wrote: Is there a way to install Asterisk from source and not stomp on your already existing Asterisk installation? I don't see a configure script and it looks like it's trying to find stuff in /etc/asterisk and in /usr/lib/asterisk and probably other places. - Jeremy Jones ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Thanks! That did it! - jmj ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] [patch] sqlite3 support for asterisk 1.2.0
Hi,
I changed cdr_sqlite so that it builds with sqlite3. I named the new module
cdr_sqlite3. It builds, but I will not be able to test it the next days.
I provide it anyway, maybe a brave heart gives me response.
Gerald
diff -Nur asterisk-1.2.0.orig/cdr/cdr_sqlite3.c
asterisk-1.2.0.sqlite3/cdr/cdr_sqlite3.c
--- asterisk-1.2.0.orig/cdr/cdr_sqlite3.c 1970-01-01 01:00:00.0
+0100
+++ asterisk-1.2.0.sqlite3/cdr/cdr_sqlite3.c2005-11-23 14:01:29.0
+0100
@@ -0,0 +1,244 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 2004 - 2005, Holger Schurig
+ *
+ *
+ * Ideas taken from other cdr_*.c files
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ *
+ * Changes for SQLite 3 by Gerald Dachs
+ *
+ */
+
+/*! \file
+ *
+ * \brief Store CDR records in a SQLite database.
+ *
+ * \author Holger Schurig [EMAIL PROTECTED]
+ *
+ * See also
+ * \arg \ref Config_cdr
+ * \arg http://www.sqlite.org/
+ *
+ * Creates the database and table on-the-fly
+ * \ingroup cdr_drivers
+ */
+
+#include sys/types.h
+
+#include stdio.h
+#include unistd.h
+#include string.h
+#include stdlib.h
+#include sqlite3.h
+
+#include asterisk.h
+
+ASTERISK_FILE_VERSION(__FILE__, $Revision: 1.11 $)
+
+#include asterisk/channel.h
+#include asterisk/module.h
+#include asterisk/logger.h
+#include asterisk/utils.h
+
+#define LOG_UNIQUEID 0
+#define LOG_USERFIELD 0
+
+/* When you change the DATE_FORMAT, be sure to change the CHAR(19) below
to something else */
+#define DATE_FORMAT %Y-%m-%d %T
+
+static char *desc = SQLite3 CDR Backend;
+static char *name = sqlite3;
+static sqlite3* db = NULL;
+
+AST_MUTEX_DEFINE_STATIC(sqlite3_lock);
+
+/*! \brief SQL table format */
+static char sql_create_table[] = CREATE TABLE cdr (
+ AcctId INTEGER PRIMARY KEY,
+ clidVARCHAR(80),
+ src VARCHAR(80),
+ dst VARCHAR(80),
+ dcontextVARCHAR(80),
+ channel VARCHAR(80),
+ dstchannel VARCHAR(80),
+ lastapp VARCHAR(80),
+ lastdataVARCHAR(80),
+ start CHAR(19),
+ answer CHAR(19),
+ end CHAR(19),
+ durationINTEGER,
+ billsec INTEGER,
+ disposition INTEGER,
+ amaflagsINTEGER,
+ accountcode VARCHAR(20)
+#if LOG_UNIQUEID
+ ,uniqueid VARCHAR(32)
+#endif
+#if LOG_USERFIELD
+ ,userfield VARCHAR(255)
+#endif
+);;
+
+static int sqlite3_log(struct ast_cdr *cdr)
+{
+ int res = 0;
+ char *zErr = 0;
+ struct tm tm;
+ time_t t;
+ char startstr[80], answerstr[80], endstr[80];
+ int count;
+ char *sqlstmt;
+
+ ast_mutex_lock(sqlite3_lock);
+
+ t = cdr-start.tv_sec;
+ localtime_r(t, tm);
+ strftime(startstr, sizeof(startstr), DATE_FORMAT, tm);
+
+ t = cdr-answer.tv_sec;
+ localtime_r(t, tm);
+ strftime(answerstr, sizeof(answerstr), DATE_FORMAT, tm);
+
+ t = cdr-end.tv_sec;
+ localtime_r(t, tm);
+ strftime(endstr, sizeof(endstr), DATE_FORMAT, tm);
+
+ for(count=0; count5; count++) {
+ if ((sqlstmt = sqlite3_mprintf(
+ INSERT INTO cdr (
+ clid,src,dst,dcontext,
+ channel,dstchannel,lastapp,lastdata,
+ start,answer,end,
+ duration,billsec,disposition,amaflags,
+ accountcode
+# if LOG_UNIQUEID
+ ,uniqueid
+# endif
+# if LOG_USERFIELD
+ ,userfield
+# endif
+ ) VALUES (
+ '%q', '%q', '%q', '%q',
+ '%q', '%q', '%q', '%q',
+ '%q', '%q', '%q',
+ %d, %d, %d, %d,
+ '%q'
+# if LOG_UNIQUEID
+ ,'%q'
+# endif
+# if LOG_USERFIELD
+ ,'%q'
+# endif
+ ),cdr-clid, cdr-src, cdr-dst, cdr-dcontext,
+ cdr-channel, cdr-dstchannel, cdr-lastapp,
cdr-lastdata,
+ startstr, answerstr, endstr,
+ cdr-duration, cdr-billsec,
[Asterisk-Users] Sip videosupport
When I enter videosupport=yes in sip.conf I can't call any SIP phone on my * (all of them are on the phone - and I directly go to voicemail). When I comment this line, everything works normal. Why * act this way? For videocalls I use eyeBeam. Thank you for your time! My sip.conf [general] externip = 111.222.333.444 fromdomain=mydomain.hr localnet=10.0.0.0/255.255.255.0 port=5060 bindaddr=0.0.0.0 context=sip srvlookup=yes dtmfmode=rfc2833 disallow=all allow=gsm allow=ulaw allow=alaw musicclass=default videosupport=yes [2026] type=friend username=2026 secret=2026 host=dynamic mailbox=2026 callerid=First Last 2026 disallow=all allow=ulaw allow=alaw allow=gsm allow=h263 allow=h263p allow=h261 canreinvite=no [2031] type=friend username=2031 secret=2031 host=dynamic mailbox=2031 callerid=First Last 2031 disallow=all allow=ulaw allow=alaw allow=gsm allow=h263 allow=h263p allow=h261 canreinvite=no -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)393447 e-mail: tparcina#lama.hr http://www.lama.hr ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] open letter (2)
may be you I agree --- Patrick [EMAIL PROTECTED] a écrit : On Wed, 2005-11-23 at 10:34 +0100, harry gaillac wrote: Advice me and I'll stop to mail my question. That almost sounds like a threat. Do you really think you motivate people to answer you this way? Since you asked this question already so many times perhaps it's time to hire a paid consultant. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] presence and Asterisk crash
Francesco Angi wrote: Hi all. I’ve got Asterisk CVS Head running on Fedora Core 3. It has been running for 4 months with no particular problem. Recently I tried to enable presence. On dialplan I added hint extensions for all my SIP users and on my Eyebeam clients (v. 1.1 3008q) I set Peer-to-Peer presence mode. Presence works right, but when an incoming or outogoing call is answered, Asterisk crashes with the following message: Ouch ... error while writing audio data: : Broken pipe Segmentation fault I tried to restart Asterisk many times but it always stop with this message. As I disable presence (on Eyebeam clients, not even in Asterisk dial plan) Asterisk stays on. Is this a bug or do I miss something with presence? There is a bug report open on this in the bug tracker. Collect some data, add a backtrace and SIP debug output up to the point where it crashes and you will help us track that bug down and kill it. Thank you for your assistance. /O ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] open letter (2)
What are your prices Harry --- harry gaillac [EMAIL PROTECTED] a écrit : may be you I agree --- Patrick [EMAIL PROTECTED] a écrit : On Wed, 2005-11-23 at 10:34 +0100, harry gaillac wrote: Advice me and I'll stop to mail my question. That almost sounds like a threat. Do you really think you motivate people to answer you this way? Since you asked this question already so many times perhaps it's time to hire a paid consultant. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to make Broadvox work with Asterisk 1.2.0
Spent quite a bit of time troubleshooting this and figured it would save someone a lot of time if this was documented (thanks to drumkilla, file and Juggie for their assistance on this as well). Been using Broadvox DIDs to receive incoming calls for over a year now with an older version of asterisk with no problems. Upgraded to asterisk 1.2.0 and incoming calls stopped working. I would see the call hit asterisk and asterisk would send a 200 OK SIP reply but Broadvox would not send an ACK. After analyzing the ethereal captures of my old system and the new system the only difference was that asterisk 1.2.0 included Max-Forwards: 70 in the SIP header when replying to Broadvox. After many hours of troubleshooting, Broadvox finally said that their switch does not support having Max-Forwards: 70 specified in the SIP headers that asterisk sends back to them so they are not able to send an ACK back. They said that I had to remove the offending statement from the SIP headers in order for the calls to work. I did so and re-tested and the calls went through fine. Here are the details of what needs to be changed to make asterisk 1.2.0 compatible with Broadvox: (assuming your source files are in /usr/src/asterisk) 1. open /usr/src/asterisk/channels/chan_sip.c for editting 2. remove each line that mentions Max-Forwards 3. save file and recompile asterisk 4. Calls from Broadvox work again They are aware that Max-Forwards is in RFC-3261 (http://www.ietf.org/rfc/rfc3261.txt). Supposedly, Broadvox is working with their vendor to update their switch to support Max-Forwards in the SIP headers. Until that happens, this is the only way to make asterisk 1.2.0 work with their equipment. Hopefully this helps! ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: I need suggestions for on equipment
Martin Joseph [EMAIL PROTECTED] wrote: On Nov 22, 2005, at 11:08 AM, Doug Meredith wrote: hugolivude [EMAIL PROTECTED] wrote: You need to be careful when buying the Linksys because version 5.0 saw a move from Linux, which runs Sveasoft's Talisman firmware, to VxWorks, which does not. Why would I care what OS an embedded device uses? Is there a difference in the externally observable behavior? Did you read the text you quoted? Ah. I did read it but I didn't understand it. When I read it I took it to mean that the Talisman firmware was part of the Linksys Linux offering. From other posts I now understand that it is in fact a replacement firmware. Doug -- Doug Meredith ([EMAIL PROTECTED]) SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and DrayTek Vigor2600VGi
richard Coco ha scritto: so the signalisation is ok. I have only problem with RTP packets (one way audio) The vigor firmware was really buggy about it. For example it was not working when externhost or externip param is set in the sip.conf file. I did notify the bug to the vigor dev team, but I don't know if they have fixed the problem yet, mine is gone really soon So an upgrade is of course necessary. Anyway you could understand the problem capping the sip packets with ethereal on both sides. I bet the ip address of the asterisk rtp box changes passing thru the vigor box and of course the device are not able to establish a right 2way audio session Let me know Sergio ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Spandsp/rxfax/txfax Asterisk 1.2stable - problems loading the modules
two things...verify the content of your /etc/ld.so.conf file must have the path included(/usr/local/lib) and recompile and install...first span and then asterisk...On 11/22/05, Dominik Simon [EMAIL PROTECTED] wrote: Hi all,today I installed asterisk 1.2stable and than spandsp-0.0.2pre21 withrxfax txfax. After I restart the asterisk and get the followingerrors:[app_rxfax.so] WARNING[6340]: loader.c:325 __load_resource: /usr/lib/ asterisk/modules/app_rxfax.so: undefined symbol: fax_set_phase_d_handlerWARNING[6340]: loader.c:554 load_modules: Loading module app_rxfax.sofailed![app_txfax.so] WARNING[6311]: loader.c:325 __load_resource: /usr/lib/ asterisk/modules/app_txfax.so: undefined symbol: fax_set_header_infoWARNING[6311]: loader.c:554 load_modules: Loading module app_txfax.sofailed!I am running Asterisk on fedora core 4 - all works great (Asterisk, app_conference and others...), but tx/rxfax failed :(Now I found the following message on http://www.asteriskguru.com/tutorials/spandsp.html:// START // 2) If you receive a message like the following:[app_txfax.so]Oct 5 12:05:24 WARNING[14665]: loader.c:314__load_resource: /usr/lib/asterisk/modules/app_txfax.so: undefinedsymbol: fax_set_header_info Oct 5 12:05:24 WARNING[14665]: loader.c:543 load_modules: Loadingmodule app_txfax.so failed!Ouch ... error while writing audio data: : Broken pipeWhen you execute the asterisk -vvvc command and the Asterisk crashes when you try to execute the safe_asterisk command, then veryprobably you have the following problem:The previously installed version of spandsp has been 0.0.3, but nowyou have installed version 0.0.2. The problem is that theinstallation of version 0.0.3 creates a symlink, which is notreplaced by installation of version 0.0.2. So the symlink points tothe library of version 0.0.3, which actually does not exist. The solution is to find the location of this symlink and to delete itmanually. Usually it is in the /usr/lib/ directory./// STOP ///-But I only have spandsl-0.0.2 installed, and the libs are in /usr/ local/lib, see:-rw-r--r-- 1 root root 946280 18. Nov 22:02 libspandsp.a-rwxr-xr-x 1 root root 822 18. Nov 22:02 libspandsp.lalrwxrwxrwx 1 root root 19 18. Nov 22:02 libspandsp.so -libspandsp.so.0.0.1lrwxrwxrwx 1 root root 19 18. Nov 22:02 libspandsp.so.0 -libspandsp.so.0.0.1-rwxr-xr-x 1 root root 738959 18. Nov 22:02 libspandsp.so.0.0.1-Does anybody have the same problems? Best regards and thx for help!Dominik Simon.___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing list Asterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk server behind NAT, and SIP clinet behind another NAT.
you need a stun server on asterisk side...I use the one that vovida.org provides...it is very easy to install and configure...On 11/23/05, jeffery chen [EMAIL PROTECTED] wrote: Asterisk server behind NAT,and SIP clinet behind another NAT.SIP.conf have set NAT=yes,SIP client can register with Asterisk server, but can not hearing anything..PLS help me, how to resolve this trouble,, As refer to the item 9http://www.voip-info.org/wiki-Asterisk+SIP+NAT+solutionsI can not register with Asterisk server too, how this happen.. _Don't just search. Find. Check out the new MSN Search!http://search.msn.click-url.com/go/onm00200636ave/direct/01/ ___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing list Asterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zyxel P2000Wv2 cannot do agent login, SJPhone work just fine?
Hi,
Problem solved sort of. For some reason I cannot get the Zyxel to work
with agentcallbackLogin when the codec is alaw, ulaw or g729 and DTMF is
rfc2833. I had to change the codec to ulaw and DTMF to inband to get it
to work. Which means the voice quality dropped some and I noticed the
echo and jitter control did not work as well, but at least now the
phones can be used to ack as an agent.
Thanks
Chuck Bunn wrote:
Hi,
Okay we have agents logging in to receive calls from a queue. Agents
logging in from a SJPhone (SIP Phone) can dial the login extension and
are asked for their 'username followed by #' and then they are asked
for their 'password followed by #' and then the system asks them what
'extension they are at followed by #'. This works perfectly. When
someone calls in the agents extensions that have logged in ring. When
someone using the Zyxel phone (by the way the latest version is a
great little phone with great clarity) calls into the agent extension
it asks for their extension as before but as soon as the user enters
the extension followed by a # the system hangs up on them, go
figure Here are my files. Oh and logging out of the agent
application works fine from SJPhone.
extensions.conf
[general]
#include macros.incl
[incoming-home]
exten = s,1,Goto(extensions-home,100,1)
exten = t,1,Goto(extensions-home,100,1)
exten = i,1,Goto(extensions-home,100,1)
[extensions-home]
include = parkedcalls
;Operator queue, Operator Console, and Receptionist Phone
exten = 100,1,Answer()
exten = 100,2,Queue(extensions-home|trn|||120)
;Office Personnel
exten = _30[0-1],1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
exten = _31[0-4],1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
exten = _399,1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
exten = _40[0-1],1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
exten = _499,1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
;Spa Personnel
exten = _500,1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
exten = _51[0-2],1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
exten = 590,1,Dial(ZAP/3,20)
;Voicemail Main
exten = 800,1,Answer
exten = 800,2,VoicemailMain
;Agent Login
exten = 801,1,AgentCallbackLogin(,,@extensions-home)
;Voice Conferencing
exten = _85X,1,Answer
exten = _85X,2,MeetMe(${EXTEN})
;exten = i,1,Voicemail(s300)
;exten = t,1,Voicemail(s300)
exten = fax,1,Dial(ZAP/4,20)
exten = fax,2,Congestion
exten = fax,102,Congestion
[local]
ignorepat = 9
exten = _9NXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})
exten = _9NXX,2,Congestion(5)
exten = _9NXX,102,congestion(5)
exten = 911,1,Dial(${OUTBOUNDTRUNK}/911)
exten = 9911,1,Dial(${OUTBOUNDTRUNK}/911
include = extensions-home
[longdistance]
ignorpat = 9
exten = _91NXXNXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})
exten = _91NXXNXX,2,Congestion(5)
exten = _91NXXNXX,102,congestion(5)
include = local
[globals]
OUTBOUNDTRUNK=Zap/G1
PSTN1=Zap/1
PSTN2=Zap/2
PHONE1=Zap/3
PHONE2=Zap/4
zapata.conf
[trunkgroups]
[channels]
echocancel=yes
echocancelwhenbridged=yes
echotraining=yes
rxgain=14.0
txgain=4.0
usecallerid=yes
hidecallerid=no
callwaiting=no
threewaycalling=no
transfer=no
immediate=no
faxdetect=both
context=incoming-home
signalling=fxs_ks
group=1
channel = 1,2
context=local
signalling=fxo_ks
group=2
channel = 3
context=longdistance
signalling=fxo_ks
group=3
channel = 4
***
queues.conf
[general]
;Operator Home
[extensions-home]
music=default
strategy=ringall
maxlen=0
member = Agent/@1
*
sip.conf
[general]
context=default
srvlookup=yes
;Zyxel - P2000WV2
[300]
context=longdistance
type=friend
username=300
secret=x
callerid=300
nat=no
host=dynamic
mailbox=300
disallow=all
allow=alaw
allow=ulaw
allow=gsm
dtmfmode=rfc2833
;Zyxel - P2000WV2
[301]
context=longdistance
type=friend
username=301
secret=x
callerid=301
nat=no
host=dynamic
mailbox=301
disallow=all
allow=alaw
allow=ulaw
allow=gsm
dtmfmode=rfc2833
.
.
.
.
;SJphone
[310]
context=longdistance
type=friend
username=310
secret=x
callerid=310
qualify=yes
nat=no
host=dynamic
mailbox=310
disallow=all
allow=alaw
allow=ulaw
allow=gsm
dtmfmode=rfc2833
;SJphone
[311]
context=longdistance
type=friend
username=311
secret=x
callerid=311
qualify=yes
nat=no
host=dynamic
mailbox=311
disallow=all
allow=alaw
allow=ulaw
allow=gsm
dtmfmode=rfc2833
...
***
agents.conf
[agents]
wrapuptime=0
musiconhold = default
updatecdr=yes
;Operator - Home
group=1
agent = 300,300,name
agent = 301,301,nam2
agent = 310,310,name3
agent = 311,311,name4
...
***
Zyxel Phone settings
*PHONE SETTINGS*
Default Voice Codec
Speaking Volume(-14~14) Listening Volume(-14~14) RTP Port
Jitter Buffer Small Medium Large Voice Frames per Packet
Small Medium Large DTMF Relay
DTMF Payload(0~127)
**
CLS Output
WHEN IT WORKS
-- Executing
Re: [Asterisk-Users] Strategy=ringall does not ring all agents.
Hi,
I have now tried other strategies including random and round robin. I am
beginning to think there is some sort of bug with Agent groups? I will
try assigning members to a queue not by their group but individually.
Thanks
Chuck Bunn wrote:
Hi,
In the queue.conf I have set the strategy set to ringall but only the
lowest
agent number ever rings??? A show agents at the CLI shows three agents
logged
in yet only the first agent ever rings. I have my agents in a group,
group 1.
queue.conf
[general]
;Operator Home
[extensions-home]
music=default
strategy=ringall
maxlen=0
context=extensions-home
member = Agent/@1
**
agents.conf
[agents]
wrapuptime=0
musiconhold = default
updatecdr=yes
;Operator - Home
group=1
agent = 300,300,name1
agent = 301,301,name2
agent = 310,310,name3
agent = 311,311,name4
agent = 312,312,name5
agent = 313,313,name6
agent = 314,314,name7
agent = 499,499,name8
;Operator - Spa
agent = 500,500,name9
agent = 510,510,name10
agent = 511,511,name11
agent = 512,512,name12
;Operator - Rest
group=2
agent = 600,600,name13
extensions.conf
[general]
#include macros.incl
[incoming-home]
exten = s,1,Goto(extensions-home,100,1)
exten = t,1,Goto(extensions-home,100,1)
exten = i,1,Goto(extensions-home,100,1)
[extensions-home]
include = parkedcalls
;Operator queue, Operator Console, and Receptionist Phone
exten = 100,1,Answer()
exten = 100,2,Queue(extensions-home|trn|||120)
;Office Personnel
exten = _30[0-1],1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
exten = _31[0-4],1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
exten = _399,1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
exten = _40[0-1],1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
exten = _499,1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
;Spa Personnel
exten = _500,1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
exten = _51[0-2],1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
exten = 590,1,Dial(ZAP/3,20)
;Voicemail Main
exten = 800,1,Answer
exten = 800,2,VoicemailMain
;Agent Login
exten = 801,1,AgentCallbackLogin(,,@extensions-home)
;Voice Conferencing
exten = _85X,1,Answer
exten = _85X,2,MeetMe(${EXTEN})
;exten = i,1,Voicemail(s300)
;exten = t,1,Voicemail(s300)
exten = fax,1,Dial(ZAP/4,20)
exten = fax,2,Congestion
exten = fax,102,Congestion
[local]
ignorepat = 9
exten = _9NXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})
exten = _9NXX,2,Congestion(5)
exten = _9NXX,102,congestion(5)
exten = 911,1,Dial(${OUTBOUNDTRUNK}/911)
exten = 9911,1,Dial(${OUTBOUNDTRUNK}/911
include = extensions-home
[longdistance]
ignorpat = 9
exten = _91NXXNXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})
exten = _91NXXNXX,2,Congestion(5)
exten = _91NXXNXX,102,congestion(5)
include = local
[globals]
OUTBOUNDTRUNK=Zap/G1
PSTN1=Zap/1
PSTN2=Zap/2
PHONE1=Zap/3
PHONE2=Zap/4
CLI Output
Starting simple switch on 'Zap/1-1'
-- Executing Goto(Zap/1-1, extensions-home|100|1) in new stack
-- Goto (extensions-home,100,1)
-- Executing Answer(Zap/1-1, ) in new stack
-- Executing Queue(Zap/1-1, extensions-home|trn|||120) in new
stack
-- outgoing agentcall, to agent '300', on
'Local/[EMAIL PROTECTED]
javascript:open_compose_win('to=300%40extensions-home-b560thismailbox=sent-mail');,1'
-- Called Agent/@1
-- Executing Macro(Local/[EMAIL PROTECTED]
javascript:open_compose_win('to=300%40extensions-home-b560thismailbox=sent-mail');,2,
stdexten|300|SIP/300) in new stack
-- Executing Dial(Local/[EMAIL PROTECTED]
javascript:open_compose_win('to=300%40extensions-home-b560thismailbox=sent-mail');,2,
SIP/300|20) in new
stack
-- Called 300
-- SIP/300-00ed is ringing
-- Agent/300 is ringing
-- SIP/300-00ed answered Local/[EMAIL PROTECTED]
javascript:open_compose_win('to=300%40extensions-home-b560thismailbox=sent-mail');,2
-- Agent/300 answered Zap/1-1
...
Thanks
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[Asterisk-Users] Re: Aastra 1.3 firmware
Lee Archer [EMAIL PROTECTED] wrote: As always right after asking it works I guess you should have asked sooner. :) Doug -- Doug Meredith ([EMAIL PROTECTED]) SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk SIP architecture question
When asterisk is setup to allow SIP users to send media end-to-end (canreinvite=yes), can cdr info still be reliable, considering one of the end-user devices could go down leaving the call open. This is assuming you are using a third party pstn and not asterisk for pstn. Does asterisk have any mechanism for detecting and disconnecting hung calls in this type of scenario? regards, David ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Strategy=ringall does not ring all agents.
Hi,
Okay I ran a test and if I define each member in a queue individually it
works. It looks like there is a bug with Agent grouping, but before I
report this as a bug I would like to know if anyone has queues working
with agent groups with Asterisk 1.2.
new queues.conf file
[general]
;Operator Home
[extensions-home]
music=default
strategy=ringall
maxlen=0
context=extensions-home
member = Agent/300
member = Agent/301
member = Agent/310
member = Agent/311
member = Agent/312
member = Agent/313
member = Agent/314
member = Agent/499
member = Agent/500
member = Agent/510
member = Agent/511
member = Agent/512
***
Thanks
Chuck Bunn wrote:
Hi,
I have now tried other strategies including random and round robin. I
am beginning to think there is some sort of bug with Agent groups? I
will try assigning members to a queue not by their group but
individually.
Thanks
Chuck Bunn wrote:
Hi,
In the queue.conf I have set the strategy set to ringall but only the
lowest
agent number ever rings??? A show agents at the CLI shows three
agents logged
in yet only the first agent ever rings. I have my agents in a group,
group 1.
queue.conf
[general]
;Operator Home
[extensions-home]
music=default
strategy=ringall
maxlen=0
context=extensions-home
member = Agent/@1
**
agents.conf
[agents]
wrapuptime=0
musiconhold = default
updatecdr=yes
;Operator - Home
group=1
agent = 300,300,name1
agent = 301,301,name2
agent = 310,310,name3
agent = 311,311,name4
agent = 312,312,name5
agent = 313,313,name6
agent = 314,314,name7
agent = 499,499,name8
;Operator - Spa
agent = 500,500,name9
agent = 510,510,name10
agent = 511,511,name11
agent = 512,512,name12
;Operator - Rest
group=2
agent = 600,600,name13
extensions.conf
[general]
#include macros.incl
[incoming-home]
exten = s,1,Goto(extensions-home,100,1)
exten = t,1,Goto(extensions-home,100,1)
exten = i,1,Goto(extensions-home,100,1)
[extensions-home]
include = parkedcalls
;Operator queue, Operator Console, and Receptionist Phone
exten = 100,1,Answer()
exten = 100,2,Queue(extensions-home|trn|||120)
;Office Personnel
exten = _30[0-1],1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
exten = _31[0-4],1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
exten = _399,1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
exten = _40[0-1],1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
exten = _499,1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
;Spa Personnel
exten = _500,1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
exten = _51[0-2],1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
exten = 590,1,Dial(ZAP/3,20)
;Voicemail Main
exten = 800,1,Answer
exten = 800,2,VoicemailMain
;Agent Login
exten = 801,1,AgentCallbackLogin(,,@extensions-home)
;Voice Conferencing
exten = _85X,1,Answer
exten = _85X,2,MeetMe(${EXTEN})
;exten = i,1,Voicemail(s300)
;exten = t,1,Voicemail(s300)
exten = fax,1,Dial(ZAP/4,20)
exten = fax,2,Congestion
exten = fax,102,Congestion
[local]
ignorepat = 9
exten = _9NXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})
exten = _9NXX,2,Congestion(5)
exten = _9NXX,102,congestion(5)
exten = 911,1,Dial(${OUTBOUNDTRUNK}/911)
exten = 9911,1,Dial(${OUTBOUNDTRUNK}/911
include = extensions-home
[longdistance]
ignorpat = 9
exten = _91NXXNXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})
exten = _91NXXNXX,2,Congestion(5)
exten = _91NXXNXX,102,congestion(5)
include = local
[globals]
OUTBOUNDTRUNK=Zap/G1
PSTN1=Zap/1
PSTN2=Zap/2
PHONE1=Zap/3
PHONE2=Zap/4
CLI Output
Starting simple switch on 'Zap/1-1'
-- Executing Goto(Zap/1-1, extensions-home|100|1) in new stack
-- Goto (extensions-home,100,1)
-- Executing Answer(Zap/1-1, ) in new stack
-- Executing Queue(Zap/1-1, extensions-home|trn|||120) in new
stack
-- outgoing agentcall, to agent '300', on
'Local/[EMAIL PROTECTED]
javascript:open_compose_win('to=300%40extensions-home-b560thismailbox=sent-mail');,1'
-- Called Agent/@1
-- Executing Macro(Local/[EMAIL PROTECTED]
javascript:open_compose_win('to=300%40extensions-home-b560thismailbox=sent-mail');,2,
stdexten|300|SIP/300) in new stack
-- Executing Dial(Local/[EMAIL PROTECTED]
javascript:open_compose_win('to=300%40extensions-home-b560thismailbox=sent-mail');,2,
SIP/300|20) in new
stack
-- Called 300
-- SIP/300-00ed is ringing
-- Agent/300 is ringing
-- SIP/300-00ed answered Local/[EMAIL PROTECTED]
javascript:open_compose_win('to=300%40extensions-home-b560thismailbox=sent-mail');,2
-- Agent/300 answered Zap/1-1
...
Thanks
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Re: [Asterisk-Users] open letter (2)
On Wed, 2005-11-23 at 14:36 +0100, harry gaillac wrote: What are your prices Don't have any since I have no idea what your problem is and how to solve it so I can't help you. Looking at the rates/pricing that were mentioned on the lists and elsewhere in the past I guess you can expect to pay around €100/hour for a good consultant. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Extension behind NAT, Asterisk on a public domain
By default AMP had NAT=yes in sip.conf, I read in some posts to change it to one, i was just trying my luck if that works. I have tried NAT=yes, The Phone gets registered, I can also make recieve calls but as soon as the call is picked I dont hear anything at both ends. Does this have anything to do with codecs? ThanksOn 11/22/05, C F [EMAIL PROTECTED] wrote: On 11/22/05, Bharath Khambadkone [EMAIL PROTECTED] wrote: Hello All,I'm fairly new to asterisk. I have read about the problems about NAT, But can't seem to find a solution. My Asterisk is on a public domain, there is no NAT or firewall in front ofIf no nat then why do you have nat=1 in sip.conf? the asteris box. I have sucessfully connected iax2 softphones was able to recieve make calls. In the same locations where I have the iax2 extensions working I have set up a a SIP softphone a SIP ATA (Sipura2002). Both teh sip phones are able to register. I can also make recieve calls but cannot hear anything after the call is answered at both ends. I'm not sure what is causing this problem. By the way I'm using SME server 7(centos 4.2)with [EMAIL PROTECTED] installed.my Sip.conf : [2008] ;(Sipura2002)username=2008type=friendsecret=2008record_out=Adhocrecord_in=Adhocqualify=noport=5060nat=1[EMAIL PROTECTED] host=dynamicdtmfmode=rfc2833context=from-internalcanreinvite=nocallerid=device 2008[2009] ;X-Lite Soft Phoneusername=2009type=friend secret=2009record_out=Adhocrecord_in=Adhocqualify=noport=5060nat=1[EMAIL PROTECTED]host=dynamicdtmfmode=rfc2833context=from-internal canreinvite=nocallerid=device 2009Thanks in advance.. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing list Asterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Server Side AgentCallbackLogin
On 11/23/05 12:00 Jason Lixfeld said the following: I'd like to not have to login, period :) I'm trying to find a way to use Queues without having to login so I don't want to have to dial an extension or anything to login. Or are you talking about having agentcallbacklogin run just before the queue is called in the dialplan? cant you use Queues with members being the channels instead of using agents ? you dont need to login that way. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Test numbers for ENUM (e164.arpa, e164.org, etc.)
On Tue, 22 Nov 2005, John Todd [EMAIL PROTECTED] wrote: I'm looking to build a decent list of test numbers which have ENUM resolution. The numbers I'm looking for should go to a recording, an echo test, or some other feature which does NOT lead to a human. These will be for manual or semi-automatic testing (i.e.: we'll test 10 times in a day, but we won't test continuously.) Any public ENUM-ish tree is fine, but I'm really shooting for e164.arpa. Not mine, so make your own mind up whether its ethical to use them, but I inadvertantly came across these recently. All go to major UK Telcos, and as anyone who has ever been an NTL customer will attest, the chance of them going straight to a human is extremely slight :) +44 800 100 152(BT business customer services) +44 800 052 2000 (NTL residential customer services) +44 800 052 8000 (NTL business customer services) +44 800 052 9000 (NTL business sales) -- Chris Hastie ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] agent transfer problem
Hi We have this problem: We have a queue with several agents logged using agentcallbacklogin If an agent receives a call and then transfer it to another agent or to another employee or to another queue, the call remains connected to the original agent. I read the archives and all the solutions I read was to use the # to transfer the calls. We are already using the # and it still doesn't free up the agent. Any idea where to look next ? Is there a debug I could activate to see where is the problem ? Could this be context related ? Patrick ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help need to reset Adit 600 for Asterisk install
Not sure but are you connecting via serial or ehternet? Seems to be the serial had a way to do this easily on bootup. Otherwise I would be interested for future reference. Carrier Access does have a good support team, just need to know your serial number. On Nov 23, 2005, at 12:13 AM, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Looking for a way to hard reset a ADIT 600 just purchased used. But it seems to have a master password already set. We've tried the front reset but maybe we don't have the right sequence of boot order. Any help would be much appreciated? - Jim ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] open letter (2)
You should read my mail so you would have an idea of my problem !!! Harry --- Patrick [EMAIL PROTECTED] a écrit : On Wed, 2005-11-23 at 14:36 +0100, harry gaillac wrote: What are your prices Don't have any since I have no idea what your problem is and how to solve it so I can't help you. Looking at the rates/pricing that were mentioned on the lists and elsewhere in the past I guess you can expect to pay around â¬100/hour for a good consultant. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Strategy=ringall does not ring all agents.
im having the same problem with ringallOn 11/23/05, Chuck Bunn [EMAIL PROTECTED] wrote:
Hi,Okay I ran a test and if I define each member in a queue individually itworks. It looks like there is a bug with Agent grouping, but before I
report this as a bug I would like to know if anyone has queues workingwith agent groups with Asterisk 1.2.new queues.conf file[general];Operator Home[extensions-home]
music=defaultstrategy=ringallmaxlen=0context=extensions-homemember = Agent/300member = Agent/301member = Agent/310member = Agent/311member = Agent/312member = Agent/313
member = Agent/314member = Agent/499member = Agent/500member = Agent/510member = Agent/511member = Agent/512***ThanksChuck Bunn wrote:
Hi, I have now tried other strategies including random and round robin. I am beginning to think there is some sort of bug with Agent groups? I will try assigning members to a queue not by their group but
individually. Thanks Chuck Bunn wrote: Hi, In the queue.conf I have set the strategy set to ringall but only the lowest agent number ever rings??? A show agents at the CLI shows three
agents logged in yet only the first agent ever rings. I have my agents in a group, group 1. queue.conf [general]
;Operator Home [extensions-home] music=default strategy=ringall maxlen=0 context=extensions-home member = Agent/@1
** agents.conf [agents] wrapuptime=0 musiconhold = default updatecdr=yes ;Operator - Home
group=1 agent = 300,300,name1 agent = 301,301,name2 agent = 310,310,name3 agent = 311,311,name4 agent = 312,312,name5
agent = 313,313,name6 agent = 314,314,name7 agent = 499,499,name8 ;Operator - Spa agent = 500,500,name9 agent = 510,510,name10
agent = 511,511,name11 agent = 512,512,name12 ;Operator - Rest group=2 agent = 600,600,name13
extensions.conf [general] #include macros.incl [incoming-home] exten = s,1,Goto(extensions-home,100,1) exten = t,1,Goto(extensions-home,100,1)
exten = i,1,Goto(extensions-home,100,1) [extensions-home] include = parkedcalls ;Operator queue, Operator Console, and Receptionist Phone
exten = 100,1,Answer() exten = 100,2,Queue(extensions-home|trn|||120) ;Office Personnel exten = _30[0-1],1,Macro(stdexten,${EXTEN},SIP/${EXTEN}) exten = _31[0-4],1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
exten = _399,1,Macro(stdexten,${EXTEN},SIP/${EXTEN}) exten = _40[0-1],1,Macro(stdexten,${EXTEN},SIP/${EXTEN}) exten = _499,1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
;Spa Personnel exten = _500,1,Macro(stdexten,${EXTEN},SIP/${EXTEN}) exten = _51[0-2],1,Macro(stdexten,${EXTEN},SIP/${EXTEN}) exten = 590,1,Dial(ZAP/3,20)
;Voicemail Main exten = 800,1,Answer exten = 800,2,VoicemailMain ;Agent Login exten = 801,1,AgentCallbackLogin(,,@extensions-home)
;Voice Conferencing exten = _85X,1,Answer exten = _85X,2,MeetMe(${EXTEN}) ;exten = i,1,Voicemail(s300) ;exten = t,1,Voicemail(s300)
exten = fax,1,Dial(ZAP/4,20) exten = fax,2,Congestion exten = fax,102,Congestion [local] ignorepat = 9 exten = _9NXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})
exten = _9NXX,2,Congestion(5) exten = _9NXX,102,congestion(5) exten = 911,1,Dial(${OUTBOUNDTRUNK}/911) exten = 9911,1,Dial(${OUTBOUNDTRUNK}/911 include = extensions-home
[longdistance] ignorpat = 9 exten = _91NXXNXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1}) exten = _91NXXNXX,2,Congestion(5) exten = _91NXXNXX,102,congestion(5)
include = local [globals] OUTBOUNDTRUNK=Zap/G1 PSTN1=Zap/1 PSTN2=Zap/2 PHONE1=Zap/3 PHONE2=Zap/4
CLI Output Starting simple switch on 'Zap/1-1'-- Executing Goto(Zap/1-1, extensions-home|100|1) in new stack
-- Goto (extensions-home,100,1)-- Executing Answer(Zap/1-1, ) in new stack-- Executing Queue(Zap/1-1, extensions-home|trn|||120) in new
stack-- outgoing agentcall, to agent '300', on 'Local/[EMAIL PROTECTED] _javascript_:open_compose_win('to=300%40extensions-home-b560thismailbox=sent-mail');,1'
-- Called Agent/@1-- Executing Macro(Local/[EMAIL PROTECTED] _javascript_:open_compose_win('to=300%40extensions-home-b560thismailbox=sent-mail');,2,
stdexten|300|SIP/300) in new stack-- Executing Dial(Local/[EMAIL PROTECTED] _javascript_:open_compose_win('to=300%40extensions-home-b560thismailbox=sent-mail');,2,
SIP/300|20) in new stack-- Called 300-- SIP/300-00ed is ringing-- Agent/300 is ringing-- SIP/300-00ed answered Local/[EMAIL PROTECTED]
_javascript_:open_compose_win('to=300%40extensions-home-b560thismailbox=sent-mail');,2-- Agent/300 answered Zap/1-1 ... Thanks ___
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Re: [Asterisk-Users] ISDN cards using CAPI interface
On Wed, 23 Nov 2005, Jesus Bermudez Riquelme - Pcmur Soluciones Informaticas wrote: Hi all, if you configure a passive ISDN card with CAPI support, does it mean that can be used with asterisk (using chan-capi, of course) ?? Yes, if you have any card/driver providing a CAPI 2.0 interface, you can use chan_capi. Armin ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP Extension behind NAT, Asterisk on a public domain
I'm pasting something from another user on this list from 14/11/05 I would recommend that you do a little research on google, voip- info.org, and the list archives. To connect to an Asterisk box that sits behind NAT, you need to forward ports 5060 and 1-2 too the asterisk box, and you need to configure the externip, localnet, and nat variables in sip.conf. audio problems are almost always due to the RTP stream (ports 1-2) not being forwarded properly, either due to the port forwarding setup or the sip.conf settings. Tom -- Tom Rymes Cascade Link Systems www.cascadelinksystems.com (603) 375-1414 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bharath KhambadkoneSent: Wednesday, November 23, 2005 9:29 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] SIP Extension behind NAT,Asterisk on a public domain By default AMP had NAT=yes in sip.conf, I read in some posts to change it to one, i was just trying my luck if that works. I have tried NAT=yes, The Phone gets registered, I can also make recieve calls but as soon as the call is picked I dont hear anything at both ends. Does this have anything to do with codecs?Thanks On 11/22/05, C F [EMAIL PROTECTED] wrote: On 11/22/05, Bharath Khambadkone [EMAIL PROTECTED] wrote: Hello All,I'm fairly new to asterisk. I have read about the problems about NAT, But can't seem to find a solution. My Asterisk is on a public domain, there is no NAT or firewall in front ofIf no nat then why do you have nat=1 in sip.conf? the asteris box. I have sucessfully connected iax2 softphones was able to recieve make calls. In the same locations where I have the iax2 extensions working I have set up a a SIP softphone a SIP ATA (Sipura2002). Both teh sip phones are able to register. I can also make recieve calls but cannot hear anything after the call is answered at both ends. I'm not sure what is causing this problem. By the way I'm using SME server 7(centos 4.2)with [EMAIL PROTECTED] installed.my Sip.conf :[2008] ;(Sipura2002)username=2008type=friendsecret=2008record_out=Adhocrecord_in=Adhocqualify=noport=5060nat=1[EMAIL PROTECTED] host=dynamicdtmfmode=rfc2833context=from-internalcanreinvite=nocallerid=device 2008[2009] ;X-Lite Soft Phoneusername=2009type=friend secret=2009record_out=Adhocrecord_in=Adhocqualify=noport=5060nat=1[EMAIL PROTECTED]host=dynamicdtmfmode=rfc2833context=from-internal canreinvite=nocallerid=device 2009Thanks in advance.. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] zyxel p2000w
Hi Chip. I am absolutely certain that the P2000w does not have a call waiting feature. You may want to check the PBX, you may have to enter in a specific key - lie * or # - in order to answer the 2nd call. We use # for a blind transfer on our asterisk - asterisk picks up the P2000w tone no problem. hope it helps thanks Chandra Mistry On 21/11/05, cp [EMAIL PROTECTED] wrote: Does anyone know is the zyxel p2000w has call waiting? I hear noise when a second call comes in but cannot find any documentation. Thanks, Chip ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Help need to reset Adit 600 for Asterisk install
Thanks Jerry, I have called Carrier Access and they can reset the password but for a considerable fee. We have serial access but after it boots it immediately asks for a username and password. We have the username but the password is not what it is suppose to be. There's a reset switch on the faceplate but I think the LOCAL SET is OFF and that is why it doesn't respond. Their manual says the Reset switch is not under the control of LOCAL SET, yet it doesn't seem to work. Well, we might not know the proper boot sequence. It contains flash memory and there is a timing that important to that reset procedure. Anyone's help is much appreciated. --Jim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jerry Jones Sent: Wednesday, November 23, 2005 7:40 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Help need to reset Adit 600 for Asterisk install Not sure but are you connecting via serial or ehternet? Seems to be the serial had a way to do this easily on bootup. Otherwise I would be interested for future reference. Carrier Access does have a good support team, just need to know your serial number. On Nov 23, 2005, at 12:13 AM, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Looking for a way to hard reset a ADIT 600 just purchased used. But it seems to have a master password already set. We've tried the front reset but maybe we don't have the right sequence of boot order. Any help would be much appreciated? - Jim ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] open letter (2)
Advice me and I'll stop to mail my question. All your base are belong to us. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] open letter (2)
New rule for email Sender = harry giallac = deleted -Original Message- From: harry gaillac [mailto:[EMAIL PROTECTED] Sent: Wednesday, November 23, 2005 8:33 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] open letter (2) may be you I agree --- Patrick [EMAIL PROTECTED] a écrit : On Wed, 2005-11-23 at 10:34 +0100, harry gaillac wrote: Advice me and I'll stop to mail my question. That almost sounds like a threat. Do you really think you motivate people to answer you this way? Since you asked this question already so many times perhaps it's time to hire a paid consultant. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ _ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Querry about the modem
Hello I have seen the article in digium site about the answering machine made using a softmodem and the zap library. I am using Fedora Core 2/3 system for doing this project. I was trying to find a PCI modem card with Intel 537 chipset. I couldn't find any model with intel 537 chipset. Can any one please get me some insight into which model I can go for. Available models here in India are, Krypton Dlink Intex Aztec ... Do any of the above modem have this chipset. Or can I use these models for this purpose. Salil G. K. kpfleming at ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Help need to reset Adit 600 for Asterisk install
Is the password limited to four digits like the Adtran 600 (I think)? Start plugging in numbers. Only 10,000 possible combinations. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Wednesday, November 23, 2005 9:59 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Help need to reset Adit 600 for Asterisk install Thanks Jerry, I have called Carrier Access and they can reset the password but for a considerable fee. We have serial access but after it boots it immediately asks for a username and password. We have the username but the password is not what it is suppose to be. There's a reset switch on the faceplate but I think the LOCAL SET is OFF and that is why it doesn't respond. Their manual says the Reset switch is not under the control of LOCAL SET, yet it doesn't seem to work. Well, we might not know the proper boot sequence. It contains flash memory and there is a timing that important to that reset procedure. Anyone's help is much appreciated. --Jim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jerry Jones Sent: Wednesday, November 23, 2005 7:40 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Help need to reset Adit 600 for Asterisk install Not sure but are you connecting via serial or ehternet? Seems to be the serial had a way to do this easily on bootup. Otherwise I would be interested for future reference. Carrier Access does have a good support team, just need to know your serial number. On Nov 23, 2005, at 12:13 AM, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Looking for a way to hard reset a ADIT 600 just purchased used. But it seems to have a master password already set. We've tried the front reset but maybe we don't have the right sequence of boot order. Any help would be much appreciated? - Jim ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Clearwire and Asterisk
I've heard that they allow only G.729 in DK -- Piotr Sobolewski [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] open letter (2)
my name is gaillac not giallac Harry --- Steve Totaro [EMAIL PROTECTED] a écrit : New rule for email Sender = harry giallac = deleted -Original Message- From: harry gaillac [mailto:[EMAIL PROTECTED] Sent: Wednesday, November 23, 2005 8:33 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] open letter (2) may be you I agree --- Patrick [EMAIL PROTECTED] a écrit : On Wed, 2005-11-23 at 10:34 +0100, harry gaillac wrote: Advice me and I'll stop to mail my question. That almost sounds like a threat. Do you really think you motivate people to answer you this way? Since you asked this question already so many times perhaps it's time to hire a paid consultant. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ _ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Help need to reset Adit 600 for Asterisk install
Does anyone know of a brute force that will work on a serial interface like hyperterminal? --Jim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Wednesday, November 23, 2005 8:22 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Help need to reset Adit 600 for Asterisk install Is the password limited to four digits like the Adtran 600 (I think)? Start plugging in numbers. Only 10,000 possible combinations. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Wednesday, November 23, 2005 9:59 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Help need to reset Adit 600 for Asterisk install Thanks Jerry, I have called Carrier Access and they can reset the password but for a considerable fee. We have serial access but after it boots it immediately asks for a username and password. We have the username but the password is not what it is suppose to be. There's a reset switch on the faceplate but I think the LOCAL SET is OFF and that is why it doesn't respond. Their manual says the Reset switch is not under the control of LOCAL SET, yet it doesn't seem to work. Well, we might not know the proper boot sequence. It contains flash memory and there is a timing that important to that reset procedure. Anyone's help is much appreciated. --Jim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jerry Jones Sent: Wednesday, November 23, 2005 7:40 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Help need to reset Adit 600 for Asterisk install Not sure but are you connecting via serial or ehternet? Seems to be the serial had a way to do this easily on bootup. Otherwise I would be interested for future reference. Carrier Access does have a good support team, just need to know your serial number. On Nov 23, 2005, at 12:13 AM, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Looking for a way to hard reset a ADIT 600 just purchased used. But it seems to have a master password already set. We've tried the front reset but maybe we don't have the right sequence of boot order. Any help would be much appreciated? - Jim ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] open letter (2)
harry gaillac wrote: my name is gaillac not giallac I think he's just dropped your maill address. Not to say that I'm suprised. You've now been asking the same questions on about 5 lists (we're all on) and it doesn't help your cause. This (and the other) lists are free resources provided by the community. Have a look on the wiki (www.voip-info.org) for Asterisk/SER consultants and if you're lucky you might find someone who isn't subscribed to the lists and therefore may help you. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk SIP architecture question
David Thomas wrote: When asterisk is setup to allow SIP users to send media end-to-end (canreinvite=yes), can cdr info still be reliable, considering one of the end-user devices could go down leaving the call open. This is assuming you are using a third party pstn and not asterisk for pstn. Does asterisk have any mechanism for detecting and disconnecting hung calls in this type of scenario? No, not accurately. Asterisk may not receive any information in this case. The best bet is that if you are doing reinvite to make an agreement with your VoIP provider to get a copy of their CDRs -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Help need to reset Adit 600 for Asterisk install
What about telnet access? If you don't know the Ethernet IP use a packet sniffer to detect it and then telnet to it. It may not be password protected. -D -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Wednesday, November 23, 2005 9:59 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Help need to reset Adit 600 for Asterisk install Thanks Jerry, I have called Carrier Access and they can reset the password but for a considerable fee. We have serial access but after it boots it immediately asks for a username and password. We have the username but the password is not what it is suppose to be. There's a reset switch on the faceplate but I think the LOCAL SET is OFF and that is why it doesn't respond. Their manual says the Reset switch is not under the control of LOCAL SET, yet it doesn't seem to work. Well, we might not know the proper boot sequence. It contains flash memory and there is a timing that important to that reset procedure. Anyone's help is much appreciated. --Jim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jerry Jones Sent: Wednesday, November 23, 2005 7:40 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Help need to reset Adit 600 for Asterisk install Not sure but are you connecting via serial or ehternet? Seems to be the serial had a way to do this easily on bootup. Otherwise I would be interested for future reference. Carrier Access does have a good support team, just need to know your serial number. On Nov 23, 2005, at 12:13 AM, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Looking for a way to hard reset a ADIT 600 just purchased used. But it seems to have a master password already set. We've tried the front reset but maybe we don't have the right sequence of boot order. Any help would be much appreciated? - Jim ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] voicemail clients
Hello to all
I have clients registered with names (joao, manuel, etc...) and clients
registered with numbers (123, 120,...).
To make the number clients receive voicemail, I have this:
exten = _X,1,Answer
exten = _X,2,Wait(1)
exten = _X,3,VoiceMail(u${EXTEN})
exten = _X,4,Playback(vm-goodbye)
exten = _X,5,Hangup
but for the name clients I need these 5 lines for each...
exten = pereira,1,Answer
exten = pereira,2,Wait(1)
exten = pereira,3,VoiceMail(u${EXTEN})
exten = pereira,4,Playback(vm-goodbye)
exten = pereira,5,Hangup
Is there any way I can solve this? making all calls that reach this
point go to the voicemail?
Thanks
Joao
___
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Re: [Asterisk-Users] open letter (2)
You've now been asking the same questions on about 5 lists (we're all on) and it doesn't help your cause. three lists. Why do you think i sent and resent my posts just for playing ? This (and the other) lists are free resources provided by the community. Have a look on the wiki (www.voip-info.org) for Asterisk/SER consultants and if you're lucky you might find someone who isn't subscribed to the lists and therefore may help you. I think Consultants have subscribed to these lists They could tell me we have the solution, here is the price Harry ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Help need to reset Adit 600 for Asterisk install
It is. Darn! --Jim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Darren Wright Sent: Wednesday, November 23, 2005 8:48 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Help need to reset Adit 600 for Asterisk install What about telnet access? If you don't know the Ethernet IP use a packet sniffer to detect it and then telnet to it. It may not be password protected. -D -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Wednesday, November 23, 2005 9:59 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Help need to reset Adit 600 for Asterisk install Thanks Jerry, I have called Carrier Access and they can reset the password but for a considerable fee. We have serial access but after it boots it immediately asks for a username and password. We have the username but the password is not what it is suppose to be. There's a reset switch on the faceplate but I think the LOCAL SET is OFF and that is why it doesn't respond. Their manual says the Reset switch is not under the control of LOCAL SET, yet it doesn't seem to work. Well, we might not know the proper boot sequence. It contains flash memory and there is a timing that important to that reset procedure. Anyone's help is much appreciated. --Jim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jerry Jones Sent: Wednesday, November 23, 2005 7:40 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Help need to reset Adit 600 for Asterisk install Not sure but are you connecting via serial or ehternet? Seems to be the serial had a way to do this easily on bootup. Otherwise I would be interested for future reference. Carrier Access does have a good support team, just need to know your serial number. On Nov 23, 2005, at 12:13 AM, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Looking for a way to hard reset a ADIT 600 just purchased used. But it seems to have a master password already set. We've tried the front reset but maybe we don't have the right sequence of boot order. Any help would be much appreciated? - Jim ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and DrayTek Vigor2600VGi
Alessio, Sergio So an upgrade is of course necessary. i have upgraded the vigor. Bad news... i am not able to register the draytek anymore. But using a XLite on my pc behind the Vigor works now fine (no one way audio). however i have an other question. I saw you put for the bindaddr same thing like 192.168.0.3. Is that the ip addr from your Asterisk? i will sniff and look wat happens... ps: Sergio, sorry for the mail... bad reply.. __ Yahoo! FareChase: Search multiple travel sites in one click. http://farechase.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] NVFaxDetect and NVBackgroundDetect on Asterisk 1.2
Justin: thank you. I had to do this for my installation as well. Ben. If you are unable to build NVFaxDetect and/or NVBackgroundDetect on Asterisk 1.2 (and/or AMP or @home Beta), make the following changes: 1) Above the following line near the top, in both files: #include asterisk/lock.h Add: #include stdio.h 2) In NVBackgroundDetect, to get rid of the trigraph warning, search for ??) and replace it with ?). 3) Rebuild Asterisk from /usr/src/asterisk with make make install. 4) Restart Asterisk with restart now from the CLI. The new release will have this modification. Justin ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Help need to reset Adit 600 for Asterisk install
Try this. http://www.thc.org/thc-hydra/ -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Wednesday, November 23, 2005 10:48 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Help need to reset Adit 600 for Asterisk install It is. Darn! --Jim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Darren Wright Sent: Wednesday, November 23, 2005 8:48 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Help need to reset Adit 600 for Asterisk install What about telnet access? If you don't know the Ethernet IP use a packet sniffer to detect it and then telnet to it. It may not be password protected. -D -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Wednesday, November 23, 2005 9:59 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Help need to reset Adit 600 for Asterisk install Thanks Jerry, I have called Carrier Access and they can reset the password but for a considerable fee. We have serial access but after it boots it immediately asks for a username and password. We have the username but the password is not what it is suppose to be. There's a reset switch on the faceplate but I think the LOCAL SET is OFF and that is why it doesn't respond. Their manual says the Reset switch is not under the control of LOCAL SET, yet it doesn't seem to work. Well, we might not know the proper boot sequence. It contains flash memory and there is a timing that important to that reset procedure. Anyone's help is much appreciated. --Jim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jerry Jones Sent: Wednesday, November 23, 2005 7:40 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Help need to reset Adit 600 for Asterisk install Not sure but are you connecting via serial or ehternet? Seems to be the serial had a way to do this easily on bootup. Otherwise I would be interested for future reference. Carrier Access does have a good support team, just need to know your serial number. On Nov 23, 2005, at 12:13 AM, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Looking for a way to hard reset a ADIT 600 just purchased used. But it seems to have a master password already set. We've tried the front reset but maybe we don't have the right sequence of boot order. Any help would be much appreciated? - Jim ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP Extension behind NAT, Asterisk on a public domain
Well, as the user stated on the original message, the asterisk server is behind a NAT and the client is also behind a NAT. if you make it work just by opening ports, let me know..I have never been able to get it to work, thats why I dont use sip, just plain iax2 for everything J Manny -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bharath Sent: Wednesday, November 23, 2005 10:08 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SIP Extension behind NAT,Asterisk on a public domain Thanks Michael, I think thats is the problem, I have opened only ports 5060-5082, I need to open 1-2 as well. I will try that and post the result when i get back home. Thanks On 11/23/05, Michael West [EMAIL PROTECTED] wrote: I'm pasting something from another user on this list from 14/11/05 I would recommend that you do a little research on google, voip- info.org, and the list archives. To connect to an Asterisk box that sits behind NAT, you need to forward ports 5060 and 1-2 too the asterisk box, and you need to configure the externip, localnet, and nat variables in sip.conf. audio problems are almost always due to the RTP stream (ports 1-2) not being forwarded properly, either due to the port forwarding setup or the sip.conf settings. Tom -- Tom Rymes Cascade Link Systems www.cascadelinksystems.com (603) 375-1414 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Bharath Khambadkone Sent: Wednesday, November 23, 2005 9:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SIP Extension behind NAT,Asterisk on a public domain By default AMP had NAT=yes in sip.conf, I read in some posts to change it to one, i was just trying my luck if that works. I have tried NAT=yes, The Phone gets registered, I can also make recieve calls but as soon as the call is picked I dont hear anything at both ends. Does this have anything to do with codecs? Thanks On 11/22/05, C F [EMAIL PROTECTED] wrote: On 11/22/05, Bharath Khambadkone [EMAIL PROTECTED] wrote: Hello All, I'm fairly new to asterisk. I have read about the problems about NAT, But can't seem to find a solution. My Asterisk is on a public domain, there is no NAT or firewall in front of If no nat then why do you have nat=1 in sip.conf? the asteris box. I have sucessfully connected iax2 softphones was able to recieve make calls. In the same locations where I have the iax2 extensions working I have set up a a SIP softphone a SIP ATA (Sipura2002). Both teh sip phones are able to register. I can also make recieve calls but cannot hear anything after the call is answered at both ends. I'm not sure what is causing this problem. By the way I'm using SME server 7(centos 4.2)with [EMAIL PROTECTED] installed. my Sip.conf : [2008] ;(Sipura2002) username=2008 type=friend secret=2008 record_out=Adhoc record_in=Adhoc qualify=no port=5060 nat=1 [EMAIL PROTECTED] host=dynamic dtmfmode=rfc2833 context=from-internal canreinvite=no callerid=device 2008 [2009] ;X-Lite Soft Phone username=2009 type=friend secret=2009 record_out=Adhoc record_in=Adhoc qualify=no port=5060 nat=1 [EMAIL PROTECTED] host=dynamic dtmfmode=rfc2833 context=from-internal canreinvite=no callerid=device 2009 Thanks in advance.. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk SIP architecture question
Matt Riddell wrote: No, not accurately. Asterisk may not receive any information in this case. The best bet is that if you are doing reinvite to make an agreement with your VoIP provider to get a copy of their CDRs Sorry, this advice is bogus :-( SIP re-INVITEs do _not_ affect the CDRs in any way, period. They only affect the media streams. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Asterisk SIP architecture question
Thanks for the information Matt! Does asterisk store any SIP dialog cdr info in mysql like Call-ID Cseq? With This info I could at least detect runaway calls and fake a BYE to the pstn gateway with an external app. regards, David On 11/23/05, Matt Riddell [EMAIL PROTECTED] wrote: David Thomas wrote: When asterisk is setup to allow SIP users to send media end-to-end (canreinvite=yes), can cdr info still be reliable, considering one of the end-user devices could go down leaving the call open. This is assuming you are using a third party pstn and not asterisk for pstn. Does asterisk have any mechanism for detecting and disconnecting hung calls in this type of scenario? No, not accurately. Asterisk may not receive any information in this case. The best bet is that if you are doing reinvite to make an agreement with your VoIP provider to get a copy of their CDRs -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Extension behind NAT, Asterisk on a public domain
Manny, Sorry if my post caused any confusion. I'm talking about 2 different locations of the server client. My Asterisk server is located at my office and is not behind a NAT or firewall. It is directly connected to my Cable modem. I'm using a Sipura2002 ATA at home. This ATA is connected to the asterisk server which is located at my office. The ATA at my home is behind a NAT. The ATA sucessfully registers and can also make recieve calls only the voice is blocked. The external ports 1-2 were not opened on my Asterisk box. Only port 5060-5082 were opened. I guess thats the reason I was not able to hear any voice. Will try that this evening and post my results. Thanks On 11/23/05, Manny A. Wise [EMAIL PROTECTED] wrote: Well, as the user stated on the original message, the asterisk server is behind a NAT and the client is also behind a NAT…. if you make it work just by opening ports, let me know..I have never been able to get it to work, that's why I don't use sip, just plain iax2 for everything… J Manny -Original Message- From: [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED]] On Behalf Of Bharath Sent: Wednesday, November 23, 2005 10:08 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SIP Extension behind NAT,Asterisk on a public domain Thanks Michael, I think thats is the problem, I have opened only ports 5060-5082, I need to open 1-2 as well. I will try that and post the result when i get back home. Thanks On 11/23/05, Michael West [EMAIL PROTECTED] wrote: I'm pasting something from another user on this list from 14/11/05 I would recommend that you do a little research on google, voip- info.org, and the list archives. To connect to an Asterisk box that sits behind NAT, you need to forward ports 5060 and 1-2 too the asterisk box, and you need to configure the externip, localnet, and nat variables in sip.conf. audio problems are almost always due to the RTP stream (ports 1-2) not being forwarded properly, either due to the port forwarding setup or the sip.conf settings. Tom -- Tom Rymes Cascade Link Systems www.cascadelinksystems.com (603) 375-1414 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Bharath Khambadkone Sent: Wednesday, November 23, 2005 9:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SIP Extension behind NAT,Asterisk on a public domain By default AMP had NAT=yes in sip.conf, I read in some posts to change it to one, i was just trying my luck if that works. I have tried NAT=yes, The Phone gets registered, I can also make recieve calls but as soon as the call is picked I dont hear anything at both ends. Does this have anything to do with codecs? Thanks On 11/22/05, C F [EMAIL PROTECTED] wrote: On 11/22/05, Bharath Khambadkone [EMAIL PROTECTED] wrote: Hello All, I'm fairly new to asterisk. I have read about the problems about NAT, But can't seem to find a solution. My Asterisk is on a public domain, there is no NAT or firewall in front of If no nat then why do you have nat=1 in sip.conf? the asteris box. I have sucessfully connected iax2 softphones was able to recieve make calls. In the same locations where I have the iax2 extensions working I have set up a a SIP softphone a SIP ATA (Sipura2002). Both teh sip phones are able to register. I can also make recieve calls but cannot hear anything after the call is answered at both ends. I'm not sure what is causing this problem. By the way I'm using SME server 7(centos 4.2)with [EMAIL PROTECTED] installed. my Sip.conf : [2008] ;(Sipura2002) username=2008 type=friend secret=2008 record_out=Adhoc record_in=Adhoc qualify=no port=5060 nat=1 [EMAIL PROTECTED] host=dynamic dtmfmode=rfc2833 context=from-internal canreinvite=no callerid=device 2008 [2009] ;X-Lite Soft Phone username=2009 type=friend secret=2009 record_out=Adhoc record_in=Adhoc qualify=no port=5060 nat=1 [EMAIL PROTECTED] host=dynamic dtmfmode=rfc2833 context=from-internal canreinvite=no callerid=device 2009 Thanks in advance.. ___--Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Asterisk SIP architecture question
Kevin, Is the CDR accounting done based on SIP signaling? If a UA is talking (RTP) to a third party PSTN gateway, isn't it at risk if say the UA loses power. How will asterisk know the call has ended if it is not involved in the media path. The idea is this.. I want to use canreinvite =yes to force users to talk end-to-end to preserve bandwidth, but I can see the potential for hung calls if asterisk never get the BYE from a UA in the event the ATA gets unplugged or somehow loses power. regards David On 11/23/05, Kevin P. Fleming [EMAIL PROTECTED] wrote: Matt Riddell wrote: No, not accurately. Asterisk may not receive any information in this case. The best bet is that if you are doing reinvite to make an agreement with your VoIP provider to get a copy of their CDRs Sorry, this advice is bogus :-( SIP re-INVITEs do _not_ affect the CDRs in any way, period. They only affect the media streams. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help on x101p disconnect when called party answers
On Mon, November 21, 2005 2:31 pm, MZ said: hi, using a multimeter i have verified that the analog line we have actually supports polarity reversal when the remote party answers and another reversal on hangup. with this i assume that i can use the kewlstart signalling so that the x101p can automatically disconnect. my problem is -- as soon as the called party answers, the call is disconnected. This is a problem here too.. Is there any way to get the x101p/TDM400p FXO ports to not disconnect when there is a polarity reversal or momentary interruption of CO battery voltage on the PSTN line ? -kim -- [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Clearwire and Asterisk
Allow only Iax in 2005/11/23, Piotr Sobolewski [EMAIL PROTECTED]: I've heard that they allow only G.729 in DK -- Piotr Sobolewski [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help on x101p disconnect when called party answers
hi, when i tried to observe the polarity reversal on my x101p (i only have this), it seems that my x101p does not care about polarity reversals but only battery interruption. my guess is that x101p is not capable of detecting polarity and i was thinking that a tdm card will solve my problem. if you have a tdm400p maybe you can try to play with the answeronpolarity/hanguponpolarity in zapata.conf. please tell me if my guess is correct that a tdm400p will solve this problem or else i'll be wasting money in buying a tdm400p. thanks On 11/24/05, Kim Culhan [EMAIL PROTECTED] wrote: On Mon, November 21, 2005 2:31 pm, MZ said: hi, using a multimeter i have verified that the analog line we have actually supports polarity reversal when the remote party answers and another reversal on hangup. with this i assume that i can use the kewlstart signalling so that the x101p can automatically disconnect. my problem is -- as soon as the called party answers, the call is disconnected.This is a problem here too.. Is there any way to get the x101p/TDM400pFXO ports to not disconnect when there is a polarity reversal or momentaryinterruption of CO battery voltage on the PSTN line ? -kim--[EMAIL PROTECTED]___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] meetme + sendtext
BJ Weschke wrote: Yes. The biggest challenge is putting together a mux device that mixes the text frames out to all of the user/channel threads in the conference. I've updated my initial patch two days ago. Now a new thread is created for each new message sent to the conference: this thread sends the message to all members of the conference, and then exits. So there is no more risk of blocking the sending channel. There is no mixing of text messages, but do we really need that ? As a side note I didn't get email from bug tracker about my update (http://bugs.digium.com/view.php?id=5808). Thanks, -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 483 527 / GSM: +689 797 527 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] LinksysOne.com (New SIP phone, and more)
Sounds like they are providing a Vonage-style service that is tied into the phones. Not sure they will sell them unlocked. Looks cool though.On 11/22/05, Lenny Tropiano / asterisk.org Mailing list [EMAIL PROTECTED] wrote:Another IP phone possibility for Asterisk.No, not the SPA941 (from the Linksys/Cisco/Sipura world)... Don't know much about it... but found this.Nothing on the datasheetsays what it'll support really.http://newsroom.cisco.com/dlls/2005/eKits/Data_Sheet_IP_Manager_Phone.pdf But I found this that also talked about it being SIP basedhttp://www.linksysinfo.org/modules.php?name=AvantGofile=printsid=438 http://www.linksysone.comEverything they want that isn't in the SPA941 ...PoE and integrated switch.Color screen.Price point $299 (estimatedlist price). Looks interesting.--Lenny TropianoE-mail: [EMAIL PROTECTED]Partner, Networking SpecialistPager:[EMAIL PROTECTED]VoIPing, LLCURL:http://www.voiping.com/PO Box 867, Cedar Park, TX 78630-0867 Mobile: 512-698-VOIP [8647]___ --Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] meetme + sendtext
On 11/23/05, Jean-Denis Girard [EMAIL PROTECTED] wrote: BJ Weschke wrote: Yes. The biggest challenge is putting together a mux device that mixes the text frames out to all of the user/channel threads in the conference. I've updated my initial patch two days ago. Now a new thread is created for each new message sent to the conference: this thread sends the message to all members of the conference, and then exits. So there is no more risk of blocking the sending channel. There is no mixing of text messages, but do we really need that ? As a side note I didn't get email from bug tracker about my update (http://bugs.digium.com/view.php?id=5808). Your latest patch is an improvement from your original one, but it still doesn't really follow the design model and creates a situation where your thread and the user's conf_run thread could be trying to get at the same channel at the same time. While you can deal with this kind of situation with channel locking, I think it'd still be cleaner to integrate the distribution of the messages into the conf_run thread that exists already for each leg in the conference. Additionally, mantis won't send you emails about updates you personally make to issues. -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] open letter
Sounds to me as what you want to do require 'a few' code changes to Asterisk. Maybe I am wrong, but this might take some work to get right. Jan harry gaillac wrote: Hello open(ser) asterisk users Here is what i expect to do : Asterisk: registrar with public ip port=5050 open(ser): outbound proxy with public ip port=5060 Asterisk don't support IM and presence so i want to use SER because of it's a good proxy: I want user agents behind nat send registration to asterisk because of it's an ipbx :-) Look at this diagram when user agent behind nat send REGISTER to ser the contact field in sip header has a private address which one is forward to asterisk for registration. When user agent are registered in asterisk AOR is sip:[EMAIL PROTECTED] ip so asterisk query sip:[EMAIL PROTECTED] behind nat (not possible). How a session between two user agents behind nat could keep in the path |register || register | agent1 asterisk| |ser/nat box || | 200 OK ||200 OK | agent2 One box --- | | | | asterisk pbx | | | | ||| | | ---- | | SER ||NAT box | private network | ---- --- Send me your questions if you don't understand what i expect to do . Harry ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] open letter
I don't know this part of Asterisk code sufficiently, but you would need to hire a programmer for this anyway. Maybe you should post a bounty into the Asterisk-Dev list I am sure one of these guys would not mind picking it up if you place a bounty. Another option that is obvious (I don't know SER) is for SER to connect to Asterisk a if it was standard SIP phones - just thinking loud... jan harry gaillac wrote: Could you tell me more please ? You understand than with host=dynamic in sip.conf asterisk use contact field in SIP HF Regards Harry --- [EMAIL PROTECTED] [EMAIL PROTECTED] a écrit : Sounds to me as what you want to do require 'a few' code changes to Asterisk. Maybe I am wrong, but this might take some work to get right. Jan harry gaillac wrote: Hello open(ser) asterisk users Here is what i expect to do : Asterisk: registrar with public ip port=5050 open(ser): outbound proxy with public ip port=5060 Asterisk don't support IM and presence so i want to use SER because of it's a good proxy: I want user agents behind nat send registration to asterisk because of it's an ipbx :-) Look at this diagram when user agent behind nat send REGISTER to ser the contact field in sip header has a private address which one is forward to asterisk for registration. When user agent are registered in asterisk AOR is sip:[EMAIL PROTECTED] ip so asterisk query sip:[EMAIL PROTECTED] behind nat (not possible). How a session between two user agents behind nat could keep in the path |register || register | agent1 asterisk| |ser/nat box || | 200 OK ||200 OK | agent2 One box --- | | | | asterisk pbx | | | | ||| | | ---- | | SER ||NAT box | private network | ---- --- Send me your questions if you don't understand what i expect to do . Harry ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk SIP architecture question
Kevin P. Fleming wrote: Matt Riddell wrote: No, not accurately. Asterisk may not receive any information in this case. The best bet is that if you are doing reinvite to make an agreement with your VoIP provider to get a copy of their CDRs Sorry, this advice is bogus :-( So how does Asterisk know that the media stream has been disconnected between the two remote hosts? -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] open letter
Could you tell me more please ? You understand than with host=dynamic in sip.conf asterisk use contact field in SIP HF Regards Harry --- [EMAIL PROTECTED] [EMAIL PROTECTED] a écrit : Sounds to me as what you want to do require 'a few' code changes to Asterisk. Maybe I am wrong, but this might take some work to get right. Jan harry gaillac wrote: Hello open(ser) asterisk users Here is what i expect to do : Asterisk: registrar with public ip port=5050 open(ser): outbound proxy with public ip port=5060 Asterisk don't support IM and presence so i want to use SER because of it's a good proxy: I want user agents behind nat send registration to asterisk because of it's an ipbx :-) Look at this diagram when user agent behind nat send REGISTER to ser the contact field in sip header has a private address which one is forward to asterisk for registration. When user agent are registered in asterisk AOR is sip:[EMAIL PROTECTED] ip so asterisk query sip:[EMAIL PROTECTED] behind nat (not possible). How a session between two user agents behind nat could keep in the path |register || register | agent1 asterisk| |ser/nat box || | 200 OK ||200 OK | agent2 One box --- | | | | asterisk pbx | | | | ||| | | ---- | | SER ||NAT box | private network | ---- --- Send me your questions if you don't understand what i expect to do . Harry ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] open letter
Could you tell me more please ? You understand than with host=dynamic in sip.conf asterisk use contact field in SIP HF Regards Harry --- [EMAIL PROTECTED] [EMAIL PROTECTED] a écrit : Sounds to me as what you want to do require 'a few' code changes to Asterisk. Maybe I am wrong, but this might take some work to get right. Jan harry gaillac wrote: Hello open(ser) asterisk users Here is what i expect to do : Asterisk: registrar with public ip port=5050 open(ser): outbound proxy with public ip port=5060 Asterisk don't support IM and presence so i want to use SER because of it's a good proxy: I want user agents behind nat send registration to asterisk because of it's an ipbx :-) Look at this diagram when user agent behind nat send REGISTER to ser the contact field in sip header has a private address which one is forward to asterisk for registration. When user agent are registered in asterisk AOR is sip:[EMAIL PROTECTED] ip so asterisk query sip:[EMAIL PROTECTED] behind nat (not possible). How a session between two user agents behind nat could keep in the path |register || register | agent1 asterisk| |ser/nat box || | 200 OK ||200 OK | agent2 One box --- | | | | asterisk pbx | | | | ||| | | ---- | | SER ||NAT box | private network | ---- --- Send me your questions if you don't understand what i expect to do . Harry ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Using transfer button in SJPhone
Hi, Does anyone know how to implement the tranfer feature (button) on the SJPhone in extension.conf Thanks ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] voicemail clients
Hello Joao, see inline.
On Wed, 2005-11-23 at 15:43 +, Joao Pereira wrote:
Hello to all
I have clients registered with names (joao, manuel, etc...) and clients
registered with numbers (123, 120,...).
To make the number clients receive voicemail, I have this:
exten = _X,1,Answer
exten = _X,2,Wait(1)
exten = _X,3,VoiceMail(u${EXTEN})
exten = _X,4,Playback(vm-goodbye)
exten = _X,5,Hangup
but for the name clients I need these 5 lines for each...
exten = pereira,1,Answer
exten = pereira,2,Wait(1)
exten = pereira,3,VoiceMail(u${EXTEN})
exten = pereira,4,Playback(vm-goodbye)
exten = pereira,5,Hangup
Is there any way I can solve this? making all calls that reach this
point go to the voicemail?
You can handle this by writing a macro.
Study this:
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Macro
Thanks
Joao
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[ sip sip:fred(at)jaolin(dot)net ]
[ web http://madpenguin.org ]
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Re: [Asterisk-Users] Multiple HFC-PCI cards in mixed modes (TE+NT) won't work!...
I also had good results with mISDN where CAPI and BRIstuff failed before. In my case just for 1 HFCPCI BRI card. The instalation ran smoothly except that in the end when I try to restart ASterisk get the message: Unable to initialize mISDN. I have the suspicion that the reason could be that no BRI line is detected as I do not have any available. Had anyone a similar problem? Jose On 23/11/05, David Woodhouse [EMAIL PROTECTED] wrote: On Tue, 2005-11-22 at 22:04 +0100, Kristof Hardy wrote: Maybe it's worth a try, using chan_mISDN (experimental, but works!).. You can find the how-to pdf (for beronet, hfc, etc.. cards) on http://www.beronet.com/downloads/. There also is an install-script that helps you through the installation, I have gotten it to work with a junghanns card and 1x HFC pci card. Didn't have a 2nd hfc around to try back then.. I you have results (good/bad) keep the list (or me) posted :) I'm using chan_misdn successfully with two HFC cards; one in NT mode and one in TE mode. -- dwmw2 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] voicemail clients
On Wednesday 23 November 2005 17:43, Joao Pereira wrote:
Hello to all
I have clients registered with names (joao, manuel, etc...) and clients
reistered with numbers (123, 120,...).
To make the number clients receive voicemail, I have this:
exten = _X,1,Answer
exten = _X,2,Wait(1)
exten = _X,3,VoiceMail(u${EXTEN})
exten = _X,4,Playback(vm-goodbye)
exten = _X,5,Hangup
but for the name clients I need these 5 lines for each...
exten = pereira,1,Answer
exten = pereira,2,Wait(1)
exten = pereira,3,VoiceMail(u${EXTEN})
exten = pereira,4,Playback(vm-goodbye)
exten = pereira,5,Hangup
Is there any way I can solve this? making all calls that reach this
point go to the voicemail?
Very ... kind of embarrassing, but interesting .
What if you assign in voicemail.conf
7373472 = 1234,pereira,[EMAIL PROTECTED]
and then exten = pereira,3,VoiceMail(u7373472)
As you understand pereira is the vanity # of 7373472
and charge pereira for having vanity # :-)
Don't forget to tell us what happened!
benchev
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Re: [Asterisk-Users] Re: Asterisk SIP architecture question
David Thomas wrote: Is the CDR accounting done based on SIP signaling? If a UA is talking (RTP) to a third party PSTN gateway, isn't it at risk if say the UA loses power. How will asterisk know the call has ended if it is not involved in the media path. The idea is this.. I want to use canreinvite =yes to force users to talk end-to-end to preserve bandwidth, but I can see the potential for hung calls if asterisk never get the BYE from a UA in the event the ATA gets unplugged or somehow loses power. That is the case in every SIP network, Asterisk is not unique in that regard. I would suggest that you could make a modification to chan_sip so that if the peer goes 'unreachable' (as determined by using qualify=yes) than any existing calls involved with that peer are immediately hung up; that would take care of this problem. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk SIP architecture question
Matt Riddell wrote: So how does Asterisk know that the media stream has been disconnected between the two remote hosts? It doesn't... nor does any other SIP softswitch. See my other reply for a possible solution. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call transfer with phones that cannot handle more than one line
Hi, Does anyone have a sample config for phones (like the Zyxel P2000wv2) that cannot handle more than one line. I have tried using # followed by the extension and nothing happens??? I have parking setup but for some reason we cannot retrieve the parked call. I call the user who the call is transfered to and they dial the parked extension in this case between 701 and 710 and nothing happens. I am just using the default feature file. *** features.conf [general] parkext = 700 parkpos = 701-720 context = parkedcalls parkingtime = 45 transferdigittimeout = 3 courtesytone = beep xfersound = beep xferfailsound = beeperr ;adsipark = yes findslot = next pickupexten = *8 featuredigittimeout = 500 [featuremap] blindxfer = #1 disconnect = *0 ;automon = *1 atxfer = *2 ** Thanks ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] astman make error
Hi all I am having an issue when trying to 'make astman'. Asterisk 1.2.0 here, from source, on debian sarge. Everything else working fine (only SIP setup anyway) deafneuron:/opt/asterisk-1.2.0/utils# make astman cc -DNO_AST_MM -c -o astman.o astman.c In file included from /usr/include/asterisk/manager.h:28, from astman.c:41: /usr/include/asterisk/lock.h: In function `ast_mutex_init': /usr/include/asterisk/lock.h:517: error: `PTHREAD_MUTEX_RECURSIVE' undeclared (first use in this function) /usr/include/asterisk/lock.h:517: error: (Each undeclared identifier is reported only once /usr/include/asterisk/lock.h:517: error: for each function it appears in.) make: *** [astman.o] Error 1 thanks for any input -- [ - ] [ idfred a.k.a chapeaurouge ] ] [ email chapeaurouge(at)madpenguin(dot)org ] [ sip sip:fred(at)jaolin(dot)net ] [ web http://madpenguin.org ] [ - ] [ GPG info ] Key fingerprint = 1674 7CBA 4F97 70D7 1A74 C7E9 333E 4335 75B1 4D02 Fetch key: gpg --keyserver pgp.mit.edu --recv-key 75B14D02 or http://madpenguin.org/pubkeys/fred_mp.key signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dial ZAP with group (g2) erroneously says call answered when it is still ringing
You can try one more thing, and that is the M option, and create a macro that announces to the user to accept the call. as documented at: http://www.voip-info.org/wiki-asterisk+cmd+dial On 11/23/05, James MacLean [EMAIL PROTECTED] wrote: Oh boy :(. As Roman politely explained in a private email... I was using ports 1 and 2 thinking they were the outbound fxs ports :(. That's it, these glasses are going, and no more testing from home :). When I switched to testing with ports 3 and 4, everything worked the same as G2. Not of course as cute as what I had hoped for when I see the local telco can do something like Dial(ZAP/g2/SIP/[EMAIL PROTECTED]) and have it wait 'til the correct phone is answered :(. Thanks to C F for the c option but my goal was to just have the 4 digit number call folks with and without SIP. I would not expect users to know to press #. I don't think dvlinedetect will quite cut it either. callprogress looked promising, but, alas, as many others have found, it hangs up after timeout seconds. I'll keep digging :). Thanks again everyone, JES James B. MacLean wrote: Hi C F, I am not well versed in this level of telephony or Asterisk, so please bare with me :). My setup is really typical. Bought the digium card with 4 ports. 2 fxs / 2 fxo. The 2 fxo's are connected directly to phones, belong to group 1 according to zapata.conf, and exist as fxoks=1-2 in /etc/zaptel.conf. The 2 fxs ports are connected to the telco, belong to group 2 according to zapata.conf, and are setup as fxsks=3-4 in zaptel.conf. Dial(Zap/1/SIP/[EMAIL PROTECTED],15,r) works as expected, Dial(Zap/2/SIP/[EMAIL PROTECTED],15,r) works as expected but: Dial(Zap/g2/SIP/[EMAIL PROTECTED],15,r) Rings once and reports answered to Asterisk. Does this support what you are explaining? I'm honestly confused by how an fxs module operates as an fxo module? Thanks for any more direction you might have, JES ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
