Re: [asterisk-users] Asterisk scalability
Hi, On Wed, 2008-01-23 at 16:03 -0500, Alex Balashov wrote: On Wed, 23 Jan 2008, Stephen Davies wrote: By the way, I have a client with a four-core Xeon box doing SIP to IAX conversion - that box can handle 1000 concurrent calls. With media passing through it? if doing conversion from sip 2 iax is pretty difficult to NOT handle media... since iax does not have RTP. regards mat ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Zonbu
Hi, On Sun, 2007-05-27 at 17:35 -0400, Nabeel Jafferali wrote: Looks like a rebadged Patton 6075 to me: http://www.patton.com/products/pe_products.asp?category=337 also patton rebrands that unit. At Cebit there was plenty of these boxes from .tw manufacturers. Matteo -- Matteo Brancaleoni RD Director Tel :+39.02.70633354 Voip :sip:[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk unable to create files, too many files open
Hi, On Wed, 2007-04-18 at 15:56 +0300, Maysara A. Abdulhaq wrote: hello, i tried to increase the number in /proc/sys/fs/file-max , which was: 203511 and file-nr was 21120 203511 so i did : echo 400176 /proc/sys/fs/file-max but it didn't help, what could possibly make this happen, and does asterisk need that huge number of files ? this machine takes less than ~40 calls at peaks! that value is a system value, not a process one. You should increase asterisk process file limit with ulimin -n before starting *. Eg ulimit -n 8192 will increase max files from the default 1024 to 8192. Greetings, matteo. -- Matteo Brancaleoni RD Director Tel :+39.02.70633354 Voip :sip:[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sending SMS from Asterisk
Hi, On Tue, 2007-02-13 at 18:04 -0700, Stephen Bosch wrote: Say I want to build an IVR application which sends an SMS message to a mobile telephone when the caller responds to a prompt in certain way. I think I can manage the part about generating the message and building something to actually send it. The part I'm foggy about is: how would I actually get the SMS message to the carrier? you can also use a gsm card. the vgsm card allows sending sms from the AMI, along with full charset support (even cirillic!), sms reports, multipage sms and so on... you can check out http://open.voismart.it/index.php/VGSM_SMS or http://open.voismart.it/index.php/VGSM_Manager_Interface matteo. -- Matteo Brancaleoni RD Director Tel :+39.02.70633354 Voip :sip:[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Detecting answer with an analogue card
Hi, On Sun, 2007-02-04 at 16:17 +0100, Stefano Corsi wrote: Hello, I have two TDM2400 card with some 40 FXS modules and 4 FXO modules. I would like use analogue lines for outboud calls. How is it possibile to detect ANSWER? you cannot. it's analogue, no signalling is done on it. unless you write dsp routines to detect the right things at the right moment :) if you want exact cdr records, you must go digital. greetings, matteo. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] issue with ivtv wctdm zaptel drivers (TDM PCI Master abort)
Hi, On Thu, 2007-01-25 at 08:03 +0100, Stefan van der Eijk wrote: Hi, I'm experiencing an issue with my x86_64 machine containing a Hauppauge PVR-500 (ivtv) and a Digium TDM400p (wctdm, part of zaptel) PCI cards. Independently of each other both cards work fine, but once the wctdm driver is loaded and mythtv tries to record something on the PVR-500 the wctdm driver freaks out. The error message is see is TDM PCI Master abort printed over and over again in the syslog. PVR cards are greedy about irq resources. They need a dedicated irq and normally the irq is held for much time. Is not a card that goes with a TDM one, since TDM cards needs a precise irq timining... Imho, you should not run both cards on same box. I had some luck doing that with a DVB-T card... since being digital cards, the amout of data transferred is lower, so can work with a TDM. But was not a hauppauge card. (I know, also the pvr500 is digital, but you have 2 tuners so double data rate and normally mpeg2 data rate in hw encoders is higher that DVB-T data rate) greetings, Matteo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Asterisk and 3PCC
uhm... On Tue, 2007-01-09 at 12:28 +0100, Gregory Duchatelet wrote: Seems that this has to be implemented by the phones, or by a B2BUA… I think that a B2BUA could be used for 3PCC, but don’t know if an open-source B2BUA exists and works with Asterisk … asterisk IS a B2BUA just my 2cents. Matteo. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] wget from within asterisk?
hi, On Fri, 2006-11-17 at 13:32 -0700, Damon Estep wrote: Basically, I want to do a wget of a URL that contains the callerID number as a variable, and assign the returned text to another variable which can be used to set the caller ID name. agi is your friend. Matteo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Vgsm driver 0.18.0 released today
Hi! for those using vGSM cards, today we released version 0.18.0, that fixes a lot of small things and implements a lot of new features, also to improve performances! In the weekend on a test customer the channel driver made 67000+ dials without a glitch! Users are encouraged to upgrade! Please take a look on http://open.voismart.it Cheers, Matteo. -- Come and visit us at VON Italy, Rome From Oct 25 to Oct 26 - Hotel Ergife Matteo Brancaleoni RD Director Tel +39.02.70633354 Sip [EMAIL PROTECTED] Iax2 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoIP GSM Gateways
Hi, On Sun, 2006-10-29 at 09:16 +0100, Michiel van Baak wrote: On 18:15, Sat 28 Oct 06, Forum wrote: I'm looking at setting up a VoIP GSM gateway to connect to my asterisk box. What experience have people on this list have with GSM gateway hardware. I have been looking at the 2N voiceblue products. Junghanns.net also has a pci card with 1, 2 or 4 simslots. That looks very good but I have no experience with it. Also we have a 2/4 gsm channels card. Many thing is that is not zaptel based and do not require any asterisk patching. Please take a look to our wiki, http://open.voismart.it were full docs are hosted. greetings, Matteo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoIP GSM Gateways
Hi, On Sun, 2006-10-29 at 13:46 +0200, Tzafrir Cohen wrote: Is vISDN (extra kernel modules, extra non-standard Asterisk channel) required? The page on vGSM there suggests it is. no, vgsm uses only a part of visdn (timer system and streamport), so you need only chan_vgsm, visdn_streamport (for audio) and visdn_timer_system for timing. Other visdn things, like chan_visdn, complex visdn pci conf etc etc is not needed. Nothing more. The card is in production since months on various systems and is running very smooth :) Matteo. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] vGSM drivers updated (0.17.2)
Hi again, On Thu, 2006-10-12 at 12:22 +0200, matteo brancaleoni wrote: Hi all, for all those using asterisk + voismart gsm cards, we have released a new package that fixes a lot of issue and add some new features. a big fix for vgsm channel driver is out on release 0.17.4, please upgrade! Greetings, Matteo -- Come and visit us at VON Italy, Rome From Oct 25 to Oct 26 - Hotel Ergife Matteo Brancaleoni RD Director Tel +39.02.70633354 Sip [EMAIL PROTECTED] Iax2 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] vGSM drivers updated (0.17.2)
Hi all, for all those using asterisk + voismart gsm cards, we have released a new package that fixes a lot of issue and add some new features. take a look to voismart open source website for it: http://open.voismart.it Greetings, Matteo. -- Matteo Brancaleoni RD Director Tel +39.02.70633354 Sip [EMAIL PROTECTED] Iax2 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] BriStuff | HFC-S | Progress() | Early B3 on incoming calls from PSTN
Hi, On Wed, 2006-08-09 at 20:50 +0200, Stefan Gofferje wrote: Hi folks, I'm currently trying to get some early audio, i.e. audio without a connection, to the caller to give some cost-free info while the alerting phase. Many (if not all) telco does not allow sending inband audio to the caller if not connected. So your caller will always get the ringtone, that's generated by the telco. Or they won't be able to sell toll free numbers :) Some telcos allow sending inband audio for few seconds on the PRI, but depends on the agreements you made. matteo. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voismart GSM - no billsecs
Hi, yes there's was a small typo that prevented answer to be detected into the channel driver. Please check the wiki (http://open.voismart.it) and try the 0-16beta1 version. matteo. -- Matteo Brancaleoni RD Director Tel +39.02.70633354 Sip [EMAIL PROTECTED] Iax2 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Digium TE110P TDM400P - always found new hardware on CentOS 4.3
Hi, On Sat, 2006-04-15 at 20:07 +0800, kevin ling wrote: Hi, I have try install the TE110P or TDM400P on HP Proliant DL380 Server. And use the AAH 2.7 distribution. When I reboot the server. The CentOS always display some hardware removed - Tiger pci card. Found new hardware - Tiger pci card. Is it the digium cards have some compatiable problems with hp DL380 server? Appreciate for any input. Thanks a lot. these cards works ok on the dl380 server. Just disable kudzu from startup sequence. (on a server you don't need it) cya, Matteo. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to get 1.2.7 asterisk
Hi, Does cvs checkout asterisk gets the later version of asterisk? I tried cvs checkout -r v1-2-7 asterisk, and didn't work for me. The only thing works is cvs checkout -r v1-2 asterisk. What exactly is version tag for version 1.2.7? Thnx Why don't you download the package from the asterisk.org website? Or checkout on the same website how to download the release with subversion. matteo ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] dual IP connections
Hi, Two asterisk servers connected across the Intenet ( in SIP or IAX mode, no matter) with both of them having static ip addresses Then I add a second link (with another provider), with another NIC at both side, and again both of them having static ip addresses. Is there a way to tell asterisk to use both of these link, i.e. doing a load balancing ? Or just better (in my case) to use only one link, and to use the second link as a backup link in the event the first link went down ? this is a routing problem, not an asterisk one. you can do some ip policy based routing , but imho if you implement this is better to have another box between the * one and the 2 isp links that do the load balance, or the switch to the bkp isp if the first one goes down. my idea is: asterisk box(one nic) - router(3 nics) - isp1 - isp2 the on router you can play with ip policy based routing or simply failover routing. cya, Matteo. -- Come to visit us @ CeBit 2006 From 9 to 15 March 2006 Hall 13 Stand no. E25/1 Matteo Brancaleoni System Administrator Tel +39.02.70633354 Sip [EMAIL PROTECTED] Iax2 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: RE : RE : [Asterisk-Users] name that vendor...
Hi, On Sat, 2005-12-31 at 17:04 +0400, Vahan Yerkanian wrote: welltech... last time i tested their fxo 4 port gateway like year ago all ports were trying to communicate using same Call-ID. we had the same issue, but the problem has been solved. just upgrade the firmware. matteo. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM400 revisions problem: Rev J not working!!
sure? have you tried latest drivers? could be simply a pci-id problem. matteo. Il giorno mar, 29/11/2005 alle 11.59 +0100, gincantalupo ha scritto: Hi, I'm trying to setup an asterisk-based PBX on a Debian Sarge distro. I'm using a K8N-E deluxe asus motherboard which gives me some problems (but I'm not sure is the motherboard causing the problem): - if I plug a TDM400 REV J, Debian cannot recognize it - if I plug a TDM400 REV E/F, everything goes well Is there anybody out there who can help me?? TIA Giorgio Incantalupo ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New ISDN architecture available for asterisk
Hi. see my comments inline. On Thu, 2005-10-20 at 11:41 +0200, Tzafrir Cohen wrote: vISDN is not based on Zaptel, libpri, chan_zap, zaphfc, qozap, etc... but has been designed from scratch to be a standard compliant EuroISDN implementation plus a channel crossconnector, plus protocol analisys support thru Ethereal, plus a ppp terminator, plus other stuff :) Ironically, the bristuff patch has currently (AFAIK, and not including vISDN which I haven't yet read much about) the best eauroisdn support and supports hfc hardware as well. yes the best that is available. but please compare it to ETSI specs. and in zaptel based... which means broken frames under load (and not only...) - Open, modular, flexible and versatile architecture - Fully GPLed What are the plans about commiting code to asterisk? openbpx support? with openpbx the support is coming. With asterisk only if digium will accept non disclaimed code, which I don't expect to happens. But if they want, they can add it. - Full support for PRI and BRI - Full support for Network and Terminal Equipment role - Traffic analisys with (patched) Ethereal - E-channel sniffing - D-channel sharing between applications (in TE-multipoint mode) - Good integration with latest 2.6 kernels and extensive usage of their newer features (e.g.: sysfs) With which versions of 2.6 should it work? With which versions was it tested? tested with 2.6.8-13 both vanilla and suse 9.3 patched and centos 4.1/2 patched. currently is developed and tested under suse 9.3 and centos 4.1. AFAIK has been tested also under fedora core 4. - Protocol stacks are implemented following the finite-state-machine models described in the ETSI specification for better compliance/debuggability. - Termination of PPP connections without exiting from the kernel - Takes advantage from HFC's framer for HDLC traffic - Preliminary/experimental support for hardware PCM bus - Preliminary support for hardware channel bridging - Support for dynamic (optionally automatic) activation/deactivation of layer2 (DLC connection) - Unintrusive with respect to the Linux kernel and Asterisk (no patches needed) Requires extra kernel modules? Extra asterisk channel modules? Oh well, I'm already building an extra zaptel-modules package... kernel modules like zaptel... the drivers :) for asterisk there's a new channel driver, which doesn't require any asterisk patching Greetings, Matteo. ___ --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] New ISDN architecture available for asterisk
Hi to all, sorry for crossposting the -dev and -user lists, but I think this could be quite interesting news for EuroISDN people, expecially BRI owners. A new ISDN architecture, called vISDN, has been developed to fully support EuroISDN protocol with HFC based cards: HFC-S PCI, HFC-4S and HFC-8S (with HFC-E1 and HFC-S USB support coming soon). vISDN is not based on Zaptel, libpri, chan_zap, zaphfc, qozap, etc... but has been designed from scratch to be a standard compliant EuroISDN implementation plus a channel crossconnector, plus protocol analisys support thru Ethereal, plus a ppp terminator, plus other stuff :) Main features: - Open, modular, flexible and versatile architecture - Fully GPLed - Full support for PRI and BRI - Full support for Network and Terminal Equipment role - Traffic analisys with (patched) Ethereal - E-channel sniffing - D-channel sharing between applications (in TE-multipoint mode) - Good integration with latest 2.6 kernels and extensive usage of their newer features (e.g.: sysfs) - Protocol stacks are implemented following the finite-state-machine models described in the ETSI specification for better compliance/debuggability. - Termination of PPP connections without exiting from the kernel - Takes advantage from HFC's framer for HDLC traffic - Preliminary/experimental support for hardware PCM bus - Preliminary support for hardware channel bridging - Support for dynamic (optionally automatic) activation/deactivation of layer2 (DLC connection) - Unintrusive with respect to the Linux kernel and Asterisk (no patches needed) Missing features: - Echo cancellation (likely going in in the next release) - Explicit Call Transfer Please note: This is a Linux-only, 2.6-only architecture which supports only EuroISDN. It is currently in beta-stage, brave testers will be more than welcome; people who want to contribute will be welcome too. You will not need to sign a disclaimer to partecipate but you will receve good and valuable consideration, plus discrete amounts of beer :^) vISDN will undergo a layer2/layer3 certification by an independent lab and will have a Declaration of conformance valid in the EU territory. AFAIK, the declaration is valid for the product as a whole, so, it is to be seen if the declaration could be extended to mixed products. Please see http://www.visdn.org/ for further informations. If you are near Milan, Italy, vISDN is live at SMAU expo. We will be happy to show the driver working live and exchange comments and ideas about it (Voismart - hall 12 booth H22) Bye! ___ --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] Reset telephone IP PHONE 106
Salve Fabio, volevo dirle che può contattare il supporto voismart a [EMAIL PROTECTED] Nel caso non avesse ricevuto risposta, mi avverta che provvederò io stesso a farle avere informazioni. (lo farei ora, ma sono sul treno e non ho accesso ai miei dati) :) greetings, Matteo brancaleoni. On Fri, 2005-10-14 at 11:15 +0200, Fabio Montemaggiore wrote: I have a telephone Voismart IP PHONE 106. I have lost the password of the telephone and therefore I am not able to set up it. How can I do to do a reset of the telephone? -- Come to visit us @ SMAU 2005 From 19 to 23 october Hall 12 Booth H22 Matteo Brancaleoni System Administrator Tel +39.02.70633354 Sip [EMAIL PROTECTED] Iax2 [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] Monitoring status of ISDN lines
this is normal for p2mp lines, expecially in Italy, where the Dchan is allowed to go down when no layer3 activity is on the BRI. Matteo. Il giorno mar, 13/09/2005 alle 19.50 +0800, Enzo Michelangeli ha scritto: When Asterisk uses an ISDN interface, it periodically sends to CLI messages such as: == Primary D-Channel on span 1 down [...] == Primary D-Channel on span 1 up Is there a simple programmatic way of capturing them for monitoring purposes? Enzo ___ --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] unamble to dialout to mobiles and others special numbers
Hi, I am able to dial out some numbers and some not. In particular it seems that i can't call mobiles and special telco numbers like the information call center, emergency numbers,... try with: pridialplan=unknown prilocaldialplan=unknown matteo -- Matteo Brancaleoni System Administrator Tel +39.02.70633354 Sip [EMAIL PROTECTED] Iax2 [EMAIL PROTECTED] ___ 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] How to restrict access to * for a specific soft/hard phone model?
I would like to allow access to * for those phones which have been tested and validated by me, e.g. calls allowed from X-lite but not from Linksys PAP2. I want to be sure that every user uses the same phone model, for example X-lite. mmmh... perhaps with sipgetheader you can get the useragent, and then drop/accept calls basing on some matching rules... Matteo -- Matteo Brancaleoni System Administrator Tel +39.02.70633354 Sip [EMAIL PROTECTED] Iax2 [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ERROR[5539]: chan_zap.c:9750 setup_zap: Unable to load config zapata.conf
hi, On Tue, 2005-06-14 at 11:35 +0200, Yousef Herzallah wrote: /etc/zaptel.conf span=1,1,0,esf,b8zs bchan=1-15,17-31# set this to 1-15,17-31 for E1 dchan=16# set this to 16 for E1 defaultzone=it loadzone=it for an E1 in italy you should use ccs,hdb3 and (normally) crc4, ie: span=1,1,0,ccs,hdb3,crc4 -- Matteo Brancaleoni System Administrator Tel +39.02.70633354 Sip [EMAIL PROTECTED] Iax2 [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ERROR[5539]: chan_zap.c:9750 setup_zap: Unable to load config zapata.conf
and also On Tue, 2005-06-14 at 11:35 +0200, Yousef Herzallah wrote: Notice: Configuration file is /etc/zaptel.conf line 0: Unable to open master device '/dev/zap/ctl' seems you have udev installed, see README.udev in zaptel src dir (and use zaptel.init to start zaptel) matteo. -- Matteo Brancaleoni System Administrator Tel +39.02.70633354 Sip [EMAIL PROTECTED] Iax2 [EMAIL PROTECTED] ___ 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] RTP Forwarding
Hi, Maybe a stun server somewhere on the internet can help you ? You could even run your own on your remote asterisk server. stun is useless if you have a simmetric nat (like any iptables firewall, cisco etc etc) matteo. -- Matteo Brancaleoni System Administrator Tel +39.02.70633354 Sip [EMAIL PROTECTED] Iax2 [EMAIL PROTECTED] ___ 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] EuroISDN Italy - quadbri - zaptel.conf - what settings work ?
You're connected to a p2mp bri, switch to bri_cpe_p2mp Matteo. Il giorno mer, 08-06-2005 alle 19:54 +0200, Robert Rozman ha scritto: Hi, I'm pulling my hair out, cause cannot connect to EuroISDN BRI in Italy with octobri card from Beronet. I use bristuff and have following zaptel.conf... # # This file is parsed by the Zaptel Configurator, ztcfg # # # First come the span definitions, in the format # span=span num,timing,line build out (LBO),framing,coding[,yellow] # # The timing parameter determines the selection of primary, secondary, and # so on sync sources. If this span should be considered a primary sync # source, then give it a value of 1. For a secondary, use 2, and so on. # To not use this as a sync source, just use 0 # loadzone=it defaultzone=it span=1,1,3,ccs,ami span=2,0,3,ccs,ami span=3,0,3,ccs,ami span=4,0,3,ccs,ami span=5,0,3,ccs,ami span=6,0,3,ccs,ami span=7,0,3,ccs,ami span=8,0,3,ccs,ami bchan=1,2 dchan=3 bchan=4,5 dchan=6 bchan=7,8 dchan=9 bchan=10,11 dchan=12 bchan=13,14 dchan=15 bchan=16,17 dchan=18 bchan=19,20 dchan=21 bchan=22,23 dchan=24 I get this on bri intense debug... Unnumbered frame: SAPI: 63 C/R: 0 EA: 0 TEI: 127EA: 1 M3: 0 P/F: 0 M2: 0 11: 3 [ UI (unnumbered information) ] 5 bytes of data Sending TEI Request ri=64864 [ fc ff 03 0f fd 60 01 ff ] Unnumbered frame: SAPI: 63 C/R: 0 EA: 0 TEI: 127EA: 1 M3: 0 P/F: 0 M2: 0 11: 3 [ UI (unnumbered information) ] 5 bytes of data Sending TEI Request ri=39384 [ fc ff 03 0f 99 d8 01 ff ] Unnumbered frame: SAPI: 63 C/R: 0 EA: 0 TEI: 127EA: 1 M3: 0 P/F: 0 M2: 0 11: 3 [ UI (unnumbered information) ] 5 bytes of data Sending TEI Request ri=38343 [ fc ff 03 0f 95 c7 01 ff ] Unnumbered frame: SAPI: 63 C/R: 0 EA: 0 TEI: 127EA: 1 M3: 0 P/F: 0 M2: 0 11: 3 [ UI (unnumbered information) ] 5 bytes of data Thanks very much in advance, regards, Rob. ___ 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 mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Digium Website Update: Asterisk Business Edition
and , what is more interesting, they've omitted any reference to digium resellers and specified only distributors :( matteo -- Matteo Brancaleoni System Administrator Tel +39.02.70633354 Sip [EMAIL PROTECTED] Iax2 [EMAIL PROTECTED] ___ 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] HP Proliant ML110 with Adaptec 2610SA and Debian
mmh I think you asked to the wrong ML, this is Asterisk, not Debian installer ML. Cya. On Wed, 2005-05-18 at 23:00 +1000, Alex wrote: Hi guys, I am trying to install Debian sarge (latest netinstall) on ML110 server with two SATA hardware mirrored drives on Adaptec 2610SA controller for use with Asterisk with no luck. Debian installer does not see the array. Any workarounds? Please help. Regards, Alex. ___ 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 -- Matteo Brancaleoni System Administrator Tel +39.02.70633354 Sip [EMAIL PROTECTED] Iax2 [EMAIL PROTECTED] ___ 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] Channel bank of E1s? (one E1 input -- 2 x E1 output)
yes, some multiplexer allows that, but they're quite expensive compared to another E1 card for asterisk. I think you'll need at least 1k $$$ for a such splitter. Matteo. Il giorno ven, 29-04-2005 alle 17:33 +0200, [EMAIL PROTECTED] ha scritto: Hi, Assume I have one E1 digium card to which I want to plug two distinct E1 PABXs , one with 15 channels and the other with 15 channels; Is there a sort of E1 multiplexer devise that allows me to plug in one hand the E1 port of the Digium card and on the other hand the two PABXs? In this same devise, I should be able to say that 15 channels need to go to first Interface and 15 other channels need to go to other interface. Or is it necessary to acquire a another E1 card although I don't need to process more channels (30 channels are ok). Any help is greatly appreciated. ___ 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 -- Matteo Brancaleoni System Administrator Tel +39.02.70633354 Sip [EMAIL PROTECTED] Iax2 [EMAIL PROTECTED] ___ 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] analog gsm router
Hi, Can you add 2 zaptel device,different ones? Like the Junghannes and a diguim analog card? Please help and advice yes you can. use fxo port cards for this. Matteo. ___ 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] wcte11xp digium card
Hi, Il giorno lun, 18-04-2005 alle 21:28 +0800, Nathaniel Angelo A. Torres (247talk) ha scritto: Hi, does anyone here tried using wcte11xp (e1) for R2 signaling. I need help because I cant make libsupertone, linunicall and libmfcr2 work. Im getting an error every time I issue the command make. Btw, the R2 variant is Philippine R2. perhaps attaching the error can be of some help? while devs listening here are very good, none of them has divination powers, till now. Matteo. -- Matteo Brancaleoni System Administrator Tel +39.02.70633354 Sip [EMAIL PROTECTED] Iax2 [EMAIL PROTECTED] ___ 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] Linux Asterisk
Hi, Il giorno gio, 07-04-2005 alle 05:19 -0400, Asterisk Pbx ha scritto: I am thinking in implementing asterisk into my buisness. I heard all sorts of good things about it. The question im asking my self is what linux distribution is best to use? Do you know what distribution they use for their asterisk training? Please search the ML. this question has been asked as many times as the number of the stars in the sky mattei ___ 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] Sangoma VS. Digium
Hi, Digium, the service is problematic. Well, I believe that Digium should services it's channels, the channels should support the resellers and the resellers should support The customers. I don't think that any company, no matter what its size or function is, could support the end users. Even the mighty ugly M$ has country based support Centers. I hate to say that, but the problem is that Digium doesn't do this. They allow resellers to do market dumping, by not imposing fixed list prices to resellers, they also compete with they're own distributors/resellers by offering the cards online and by offering services directly to end users. In this way they're destroying they're own reseller network and there's no commercial gain into supporting the end user (as resellers). Sangoma doesn't do that. they don't sell directly, thus allowing resellers to have a money gain and pay the time to support the end user. again, I hate to say that, but is a common pow. I hope that digium will change their mind in the way they sells hw/services. Matteo ___ 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] Sangoma VS. Digium
Hi, Il giorno gio, 07-04-2005 alle 13:02 +0200, [EMAIL PROTECTED] ha scritto: Digium do sell online and so many other of their resellers do. The important point is that they don't sell lower cost than their resellers, which is the case. Please find an hardware producer that sells directly to endusers, when they have also distributors/resellers. The way is: if you have resellers, sell through them. if not directly to end user. Reseller added value is find customers and retail locally in his place with local variables of config, ...etc. They are the ones to find customers and to make sure they bring added value. Yes of course. but they're sure that the customer will buy from them. normally the user will buy directly from the hw maker (and this's ok) if the hw maker allows that, since in this way the user thinks that going directly to the manufacturer they'll have better support and better price. I know that is can not be the real truth, but is how's perceived from an enduser pow. We're Digium resellers, but some .it people buy the card from other countries (because not imposing list prices allows resellers to do market dumping) or even direlcty from Digium. And we apply the very same Digium list price. and the import taxes are payed by our reseller discount. So when the enduser buys directly from .usa, they will pay list price plus taxes, so more than our final price. But this is not considered, seems. I don't know what's unusual in this approach? everything. Matteo ___ 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] Call Interception
Hi, Thanks answering me, that works with the *8 (and *02 th e pattern in my company works too) but there is a problem : how do you select the phone ringing to pickup ? For example phones 23 and 24 are ringing ; I'm 25 (same pickupgroup as 23 and 24 callgroup), How do I decide either to take the 23 or 24 ? Seems the *8 takes the first arrived call. Any idea ? Asterisk does not have a directed call pickup implemented within it. Not sure how one would try to implement that, but a guess would be that it would require an external script or app of some sort. patch with bristuff and you'll have it. Directly pickup single channels (basing on exten, for example) :) Matteo. ___ 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] MWI for SER and Asterisk - ast_data vs realtime
Hi, ast_data is replaced by ARA (Asterisk RealTime Architecture). That is why you can't compile ast_data on HEAD. I'm currently testing ast_data on HEAD. the patches applies and compiles :) ARA is something different from ast_data: * ast_data is real realtime: for example extensions.conf is looked up from DB in realtime without reload. With ARA you can only: * load the whole dialplan from DB (ARA static), (not realtime) an apply mods with reload * make it dynamic with switch statements. ast_data is like having the whole dialplan in DYNAMIC mode. This can be desiderable in certain situations (ie quickly changing dialplan) * ast_data iaxfriends sipfriends is like ARA dynamic sipiax friends, but with a lot of params in addition for the peer. just look the patches. now I'm starting a project where I need this stuff. but since I need it NOW! I think I'll go with asterisk stable + ast_data, even if ast_data works on HEAD. But on head we're having a lots of commits that needs testing, so I don't feel it will be very production ready (but this's the reason of HEAD, so is ok) Matteo ___ 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] MWI for SER and Asterisk - ast_data vs realtime
Hi, ast_data is replaced by ARA (Asterisk RealTime Architecture). That is why you can't compile ast_data on HEAD. I'm currently testing ast_data on HEAD. the patches applies and compiles :) One note: you need to use the SVN dev version for that. Matteo ___ 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] Call Interception
Hi, It's working fine, although I'm not sure if it comes with asterisk or with bristuff ... bristuff Matteo ___ 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] Web interface for realtime Mysql friends/peer
phpmyadmin :) Matteo. Il giorno mer, 06-04-2005 alle 20:05 +1100, Laurent Foulonneau ha scritto: Hello list, Does anyone know about a web/php interface to deal with users in Realtime's Mysql database (sipusers and sippeers tables) ? Thanks in advance Laurent ___ 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
Hi, first of all: DON'T crosspost. chances to be ignored and/or flamed are very high. then... I?m a telecommunication engineering student. I?m working on my degree thesis, it?s about Astrerisk . My goal is to estimate the performance of a hybrid platform for the Volp. I?m looking for documentation about: ? Architecture ? Tools for the performances? analysis (to analyse performances) ? Informations about the scheduler ? Informations about the transcoding, to understand how the Volp Protocol (Sip,H.323,IAX) interact read the code guy, is opensource :) if you need to understand docs are never enough, so if you're going for a real degree, only reading the code will give you degree-level results... and also if you go deep enough perhaps you'll have some very useful suggestions for the * community... matteo. -- Matteo Brancaleoni Espia System Administrator http://www.espia.it This message was sent using IMP, the Internet Messaging Program. Service is provided by Espia - Emmegi Srl - http://www.espia.it. ___ 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] Zap timing device
Hi, Can someone tell me if the timing device is needed for voicemail and other applications too?. i'm sure that searching on google and/or voip-info.org can lead to an answer. btw, the answer is no. only meetme and iax truking needs a timing device. -- Matteo Brancaleoni System Administrator Tel +39.02.70633354 Sip [EMAIL PROTECTED] Iax2 [EMAIL PROTECTED] ** Visit us at CeBIT 2005 - Hall 13,Booth D51 ** ___ 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] B2BUA
agi + dial option T is your friend. Matteo. Il mar, 2005-01-18 alle 15:24, Joao Pereira ha scritto: Hello to all Im using SER as SIP registrar and Asterisk as GW and billing system but I m not sure if Asterisk can interupt calls when a client is out of credit. Is there any way of doing it or I need to use B2BUA ? Thanks Joao Pereira ___ 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 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] not sharing IRQ's
Hi, Il giorno mer, 12-01-2005 alle 20:29 +0330, Paradise Dove ha scritto: just to make sure: when i have zaptel devices on my box and i also use meetme and iax2, do i need to have USB device enabled and it's modules loaded? no, no, no just usbcore+uhci loaded + ztdummy Matteo. -- Matteo Brancaleoni System Administrator Tel +39.02.70633354 Sip [EMAIL PROTECTED] Iax2 [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Asterisk and Capi
Hi, [app_capiCD.so]Dec 23 19:21:45 WARNING[1076850816]: loader.c:242 ast_load_resource: /usr/lib/asterisk/modules/app_capiCD.so: undefined symbol: capidebug Dec 23 19:21:45 WARNING[1076850816]: loader.c:423 load_modules: Loading module app_capiCD.so failed! in * modules.conf, be sure to have chan_capi.so=yes under the global section and load = chan_capi.so under the modules section. This permits early load on capi driver and export symbols to other applications, like app_capiCD.so Matteo. P.S. and after that go with the sources, it the only way to understand how the things works! -- Matteo Brancaleoni System Administrator Tel +39.02.70633354 Sip [EMAIL PROTECTED] Iax2 [EMAIL PROTECTED] ___ 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] TE110P + Asterisk
Ciao, l gio, 2004-12-02 alle 14:10, Leonardo Tramontina ha scritto: snip -- Registered channel 23, PRI Signalling signalling Dec 2 11:07:15 ERROR[5209]: chan_zap.c:6447 mkintf: Channel 24 is reserved for D-channel. Dec 2 11:07:15 ERROR[5209]: chan_zap.c:9274 setup_zap: Unable to register channel '1-15' Dec 2 11:07:15 WARNING[5209]: loader.c:396 ast_load_resource: chan_zap.so: load_module failed, returning -1 perhaps you should move the jumper to the E1 (closed) position? Matteo. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] wiki down ?
Hi, Il giorno lun, 22-11-2004 alle 08:49 -0500, Jason p ha scritto: Fatal error: Unknown function: mssql_get_last_message() in /var/www/html/tikiwiki-1.8.2/lib/adodb/drivers/adodb-mssql.inc.php on also here...perhaps they're switching away from mssql ? :) Matteo -- Matteo Brancaleoni System Administrator Tel +39.02.70633354 Sip [EMAIL PROTECTED] Iax2 [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] SysMaster and GPL Violation
Hi, Il ven, 2004-11-12 alle 07:16, Jeremy McNamara ha scritto: Brian West wrote: So all you Sysmaster owners run strings on the 'voipgw' binary that runs on those boxes and you'll see that its asterisk. If you have doubts I'll post more proof. snip ... I too demand sysmaster either pay Digium for a non-gpl license or publicly admit the fact that they have repackaged Asterisk and contribute enhancements to Asterisk back to the GPL. *if they have made any enhancements* :) Matteo ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] SysMaster and GPL Violation
Hi, If the facts are really obvious that they are hiding Asterisk in their system best would be to make it public I think. That means informing the well-known online magazines and see what happening ;-) what's about slashdot? Matteo -- Matteo Brancaleoni System Administrator [EMAIL PROTECTED] EspiA Srl - e*solution provider Via Pascoli, 37 20129 Milano - Italy SIP:[EMAIL PROTECTED] Tel. +39 0270633354 Fax. +39 0245487890 IAXTEL: 17005662458 http://www.espia.it ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Pause during dial
Hi, Il mer, 2004-11-10 alle 17:27, Henry Devito ha scritto: Is there a way to put pauses in a dial string? I need * to dial a number then pause for 6 seconds and dial a second string of numbers. search the list. This question has been answered tons of time before. Matteo. -- Matteo Brancaleoni System Administrator [EMAIL PROTECTED] EspiA Srl - e*solution provider Via Pascoli, 37 20129 Milano - Italy SIP:[EMAIL PROTECTED] Tel. +39 0270633354 Fax. +39 0245487890 IAXTEL: 17005662458 http://www.espia.it ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] NoOp
Hi,
Sometimes I see in a context NoOp
What is the purpose of NoOp (no operation) if it does nothing?
Exactly that. Doing nothing :)
btw, noop could be a placeholder for future instructions,
or if you need to delete an application from the dialplan,
saves you from renumbering the priorities.
Or also it helps for debugging the dialplan,
since it can print vars. For example,
exten = s,1,Noop(${CALLERIDNAME})
will print on the console the value
of the var CALLERIDNAME.
Matteo.
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Re: [Asterisk-Users] Reload Asterisk from php or perl script
Hi,
I am looking for a basic script that can reload asterisk from
php or perl via a web browser.
I have tried exec( asterisk -rx reload ) and shell_exec( same cmd )
with php but there seems to be a permission issue with asterisk that
stops these working. I was just wondering if anyone has a way around
this with perl or php.
besides I prefer to use the manager, cause is more secure,
easy, etc, another way to reload from php is to call the
script with a wrapper in perl, like:
test.php is the script that does fancy things and contains
something like asterisk -rx reload somewhere,
and /or writes * config files, blah blah...
the test perl script would be something like:
#** cut here
#!/usr/bin/perl
# Perl wrapper to execute a PHP script setuid
# Requires PHP CLI
use File::Basename;
# Make
UID = EUID (so that PHP can run system()s and execs() setuid)
$ = $;
# Set
this to the path, so that we can't get poisoned
$ENV{'PATH'} = /var/lib/asterisk/scripts;
$ENV{'BASH_ENV'} = /var/lib/asterisk/scripts;
# Open
the PHP script
$data = basename($0);
if
($data =~ /^([EMAIL PROTECTED])$/) {
$data = $1; # $data now untainted
} else {
die Bad data in $data;# log this somewhere
}
system($data..php);
#** cut here
and call /var/lib/asterisk/scripts/test
btw, the manager is better :)
Matteo.
--
Matteo Brancaleoni
System Administrator
[EMAIL PROTECTED]
EspiA Srl - e*solution provider
Via Pascoli, 37
20129 Milano - Italy
SIP:[EMAIL PROTECTED]
Tel. +39 0270633354
Fax. +39 0245487890
IAXTEL: 17005662458
http://www.espia.it
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[Asterisk-Users] Status of conference calls at Astricon ?
On late august, there was a thread about setting up some meetme conferences to be able to follow Astricon remotely. This indeed could be nice for those that can't attend for various reason. And of course is a demonstration of Asterisk capabilities... :) (Astricon without a remote conference for guest is like a big it expo without internet connections...) I have some bandwidth here, so can set up quickly a server for .it conference termination... so, bring on and demostrate to the world what asterisk can do! Matteo :) -- Matteo Brancaleoni System Administrator [EMAIL PROTECTED] EspiA Srl - e*solution provider Via Pascoli, 37 20129 Milano - Italy SIP:[EMAIL PROTECTED] Tel. +39 0270633354 Fax. +39 0245487890 IAXTEL: 17005662458 http://www.espia.it ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Status of conference calls at Astricon ?
/me too this morning was all okie, now I can't connect. I have an asterisk server ready for replicate the conference here in .it, as soon as the link will be up with someone, I'll post the IAX2 url Matteo. -- Matteo Brancaleoni System Administrator [EMAIL PROTECTED] EspiA Srl - e*solution provider Via Pascoli, 37 20129 Milano - Italy SIP:[EMAIL PROTECTED] Tel. +39 0270633354 Fax. +39 0245487890 IAXTEL: 17005662458 http://www.espia.it ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Issue with TE405P and Adaptec U160 SCSI
are the scsi and te405p irq shared? te405p hates shared irqs... matteo. -- Matteo Brancaleoni System Administrator [EMAIL PROTECTED] EspiA Srl - e*solution provider Via Pascoli, 37 20129 Milano - Italy SIP:[EMAIL PROTECTED] Tel. +39 0270633354 Fax. +39 0245487890 IAXTEL: 17005662458 http://www.espia.it ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Non standard usage of X100P card.
Hi Il ven, 2004-07-30 alle 11:18, Dmitry Sergeev ha scritto: I have two X100P card in my box. I want to connect regular phone (not the phone line!) to one of thse cards. Does anybody think about the same? no. is completely different. I don't really want an expensive solution buying additional card with FXS port, I prefer to make something by myself. It'll be great if somebody can point me to technical materials or show electric scheme of such converter. I believe it should be rather simple. the material needed and the time (assuming that's possible) will be much more that the price of a single fxs card... first of all : you'll need to supply power to the line from the card then you must have a ringer to ring the phone (eh, something like 60/70 volts...) then... you must say the card to activate the ringer when needed ... but perhaps this is the simplest step. it's that worthwhile ? -- Matteo Brancaleoni System Administrator [EMAIL PROTECTED] EspiA Srl - e*solution provider Via Pascoli, 37 20129 Milano - Italy SIP:[EMAIL PROTECTED] Tel. +39 0270633354 Fax. +39 0245487890 IAXTEL: 17005662458 http://www.espia.it ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Poopy errors on quad wcfxo
Hi all, I'm experiencing problems with the TDM card with 4 fxo modules. on all tests, if the cards has 4 modules, I get poopy kernel messages on the card. The card works for sometime,then hangs and a asterisk restart must be done, along with kern modules unload/reload . if I remove the first module, the card works without problems at all on the remaining 3 modules. using latest zaptel cvs. anyone is experiencing that or have a workaround ? thanks a lot, Matteo -- Matteo Brancaleoni [EMAIL PROTECTED] Espia - Emmegi Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Multi process of *
Hi. Johnson-Perkins, Robert wrote: If you are just doing VoIP (i.e. no FXO/FXS Cards involved) you should be able to run up multiple virtual copies of Linux * in VMWare or Virtual PC. Though I guess you would need a pretty pokey machine User Mode Linux is way better for that use, much more efficient. Matteo. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] E1 channel bank problem
Hi all. I have and E1 channel bank from Loop Telecom. there's a little issue with it, I cannot ring the phones on fxs interface, but can connect without issue them. What happens: I dial the phone on port 1, asterisk says Zap/1 is ringing, but the phone on the analog port doesn't ring. but if I take off hook the ringed phone, asterisk detects the answer at they're bridged correctly. also I can flash transfer without probs. only ring doesn't work. doing the ring test from the channel bank test menu, is all ok: the phones ring without issues. zaptel.conf says: span = 1,1,0,cas,hdb3,crc4 fxoks = 1-31 loadzone = us defaultzone = us zapata.conf is simply transfer=yes echocancel=yes threewaycalling=yes signalling=fxo_ks context=interni channel=1-31 any hint on where I can search for problems? -- Matteo Brancaleoni [EMAIL PROTECTED] Espia - Emmegi Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Old sound in new call.
this is an old problem. there's a kernel patch for that. search the mailing lists archives. Matteo. Il mer, 2004-05-19 alle 13:42, Michael Ljtnant ha scritto: Hi, I have a problem that I just can't figure out how to solve. I start *, dial it using a ISDN phone over PSTM, to a Hisax card installed in * I get the demo-greeting, listen for a few seconds and hang up. I dial it again, but this time the first second is sound from where the previous call ended, then the greeting starts as it should. Right now I have removed all codecs but codec_gsm.so and format_gsm.so, recompiled the kernel over and over, but I just can't figure it out. Any one got some ideas? Modem.conf: [interfaces] context=incomming driver=i4l dialtype=tone mode=immediate language=en type=autodetect group=1 msn=39660425 incomingmsn=39660425,39660426 device = /dev/ttyI0 device = /dev/ttyI1 Extensions.conf: [default] include = incomming [incomming] exten = 39660425,1,Wait,1 exten = 39660425,2,Answer ; Answer the line exten = 39660425,3,BackGround(demo-congrats) exten = 39660425,4,Hangup System Info: 1 x Intel(R) Pentium(R) 4 CPU 2.53GHz SMP Motherboard. 512 MB Ram Linux-2.4.26 (Compiled as SMP) Latest Asterisk Devel CVS version mpg123 Version 0.59s-mh4 Hisax card: Cologne Chip Designs GmbH ISDN network controller [HFC-PCI] (rev 2). Wildcard TDM400P Wildcard X100P -- Matteo Brancaleoni [EMAIL PROTECTED] Espia - Emmegi Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] * ISDN-BRI-PTP DID ISDN4Linux does not show incoming number
and why not getting a quadbri card from the same author of chan_capi ? matteo. Il ven, 2004-05-07 alle 11:05, Andreas Frackowiak ha scritto: Hallo Felix, it seems that the FAQ only describes windows co. Just try to use the capi driver, I guess you would get much more support for capi here... Well now I am sure: The AVM-Fritz-CAPI does not work with PTP. o I have tried it and it doesn't work o I asked AVM and they answered that the Fritz CAPI-Software (Windows + Linux) does not support DDI/PTP-Mode. o I found a lot of messages in old archives of this list and the i4l-list which also say that PTP with Fritz CAPI does not work. Also mISDN (ISDN4Linux successor with CAPI20) maybe will support P2P with Fritz Card sometime, but not today. And so it seems that my problem between ISDN4Linux and the chan_modem_i4l driver remains an unsolved mystery. So maybe I have to buy an AVM B1 or C2 card to circumvent this problem or use something else than asterisk. thanks and regards Andreas -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andreas Frackowiak Sent: Wednesday, May 05, 2004 8:08 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] * ISDN-BRI-PTP DID ISDN4Linux does not show incoming number Hi Felix, I am using Asterisk on a DSS1 ISDN-BRI with ISDN4Linux (and a Fritz Card PnP). The ISDN-BRI is in PTP-Mode (Point to Point german: Anlagenanschluss) which is enabled within I4L with hisaxctrl fcpcipnp0 7 1. are you shure, that the capi does not support PTP? I have an AVM C4 card, but it should be the same with the fritz.. Well, I am not sure, but AVM says in: http://www.avm.de/de/Service/FAQs/FAQ_Sammlung/2671.php3 that only the B1-family of cards and the C2 and C4 Controllers support PTP. I would be very happy if someone has a Fritz with CAPI working with a PTP und could proove that I am wrong. I also would be very happy if someone could help me with the original question, why I4L does not give the called number / MSN to Asterisk (and help me fix it, of course :) Thanks Andreas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Matteo Brancaleoni Espia System Administrator Email : [EMAIL PROTECTED] Web : http://www.espia.it Phone : +39 02 70633354 - ext 201 IAX(2): [EMAIL PROTECTED] - ext 201 Iaxtel: 1-700-56-62458 - ext 201 SIP : [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] mpg123 versions ?
Hi. What versions does everyone use without problems. 0.59r is PERFECT 0.59r here. all ok. matteo. -- Matteo Brancaleoni Espia System Administrator Email : [EMAIL PROTECTED] Web : http://www.espia.it Phone : +39 02 70633354 - ext 201 IAX(2): [EMAIL PROTECTED] - ext 201 Iaxtel: 1-700-56-62458 - ext 201 SIP : [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] module help?
Hi. flame mode on Need some help with modules.conf, and basic RH9 linux skills. perhaps wrong list? see linux kernel howto... I've installed the new TDM04B 4-port FXO card and its working. After a reboot, when I do lsmod I see the wcfxo module but not the wcfxs even though both are listed modules.conf. If I modprobe wcfxs, then lsmod has both modules showing. why you need wcfxs on a quad-fxo ? The wcfxs module is the last one in the modules.conf. Is the order of entries sensitive in modules.conf? modules.conf != loaded modules. as the name suggest, it contains only configuration params for modules Do I need to be concerned with wcfxs not showing before starting asterisk? Any suggestions? sure. learn something more about kernel, modules and what is modules.conf bug us with asterisk related questions, not with what-are-kernel-modules? questions. /flame mode off -- Matteo Brancaleoni Espia System Administrator http://www.espia.it This message was sent using IMP, the Internet Messaging Program. Service is provided by Espia - Emmegi Srl - http://www.espia.it. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 or voicemail2?
Scrive Paul Mahler [EMAIL PROTECTED]: I'm using the stable branch. Is voicemail or voicemail2 deprecated? RGH!!! ages passed when voicemail was sent to /dev/null and voicemail2 moved to voicemail... current voicemail is the old voicemail2 voicemail doesn't exist any more. perhaps voicemail2 exists only as an alias to voicemail to make the transition smoother Matteo. -- Matteo Brancaleoni Espia System Administrator http://www.espia.it This message was sent using IMP, the Internet Messaging Program. Service is provided by Espia - Emmegi Srl - http://www.espia.it. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] problems with bri-stuff.0.0.2rc20a
immediate=no is in the right position into zapata.conf? ie before the channel=XX you're picking up? matteo. Scrive FastJack [EMAIL PROTECTED]: hi everybody, just upgraded my bri-stuff driver to 0.0.2rc20a. now i have a strange problem :-( i have immediate = no but when i pickup the phone i get : *CLI == D-Channel on span 1 up -- Extension 's' in context 'default' from '6294094' does not exist. Rejecting call on channel 2, span 1 i have started asterisk with -vvc so there should be a debug message if immediate mode was on. maybe anyone (klaus-peter) can help. i'm using a hfc-card in nt-mode. i'm not 100% shure but i think that my phone is using uk-tones (ring ...) since the update but all language-settings are nl. looking forward to get some help ;) thorsten ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Matteo Brancaleoni Espia System Administrator http://www.espia.it This message was sent using IMP, the Internet Messaging Program. Service is provided by Espia - Emmegi Srl - http://www.espia.it. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Cisco 7960 SIP Firmware
you should get that from the seller of the phones, they must have a CCO login with donwload privs and give you the firmware. but if u bought them used, that's another story It's not legal to share cisco firmware without authorization... Matteo. Il ven, 2004-04-23 alle 10:38, Johnson-Perkins, Robert ha scritto: I have just got 3 Cisco 7960 phones which I would like to connect to Asterisk... However they seem to have v3 SCCP firmware. I have tried numerous links to the Cisco Website but unable to get the SIP firmware. Has anyone managed to get a service contract or an account with download privileges? Ideally I would like to upgrade to 6.3 SIP; though it seems I might need to upgrade via v3 or v4? Any idea where I might find copies? robert AT johnson-perkins DOT com PLEASE READ: The information contained in this email is confidential and intended for the named recipient(s) only. If you are not an intended recipient of this email you must not copy, distribute or take any further action in reliance on it and you should delete it and notify the sender immediately. Email is not a secure method of communication and Nomura International plc cannot accept responsibility for the accuracy or completeness of this message or any attachment(s). Please examine this email for virus infection, for which Nomura International plc accepts no responsibility. If verification of this email is sought then please request a hard copy. Unless otherwise stated any views or opinions presented are solely those of the author and do not represent those of Nomura International plc. This email is intended for informational purposes only and is not a solicitation or offer to buy or sell securities or related financial instruments. Nomura International plc is regulated by the Financial Services Authority and is a member of the London Stock Exchange. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Matteo Brancaleoni Espia System Administrator Email : [EMAIL PROTECTED] Web : http://www.espia.it Phone : +39 02 70633354 - ext 201 IAX(2): [EMAIL PROTECTED] - ext 201 Iaxtel: 1-700-56-62458 - ext 201 SIP : [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CAPI and Extensions.conf Security problem
ever heard of a 'correct dialplan' ? perhaps there's some bug in your context/extensions logic that let this happens. better review it :) Matteo. Il ven, 2004-04-23 alle 11:20, Ignace CARIA ha scritto: Hi, I've installing a AVM Fritz Card in my ASterisk Box I've configured everything and its running perfectly. The problem is that everybody is allow to call through it. Explaination: All users registered in Asterisk can make a call towards the ISDN network But, everybody from the Internet, knowing the extension of CAPI in the dialplan, can call through my Asterisk to any phone number Heellp mmm please ! Thanks Ignace ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Matteo Brancaleoni Espia System Administrator Email : [EMAIL PROTECTED] Web : http://www.espia.it Phone : +39 02 70633354 - ext 201 IAX(2): [EMAIL PROTECTED] - ext 201 Iaxtel: 1-700-56-62458 - ext 201 SIP : [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Re[2]: [Asterisk-Users] dtmf for public telephony access
depends on the device you're using, if are supported or not. i feel very confortable with INFO method, since is a sip message and can be easily debugged :) Il gio, 2004-04-15 alle 09:45, Alessio Focardi ha scritto: Grazie Matteo, I looked in wiki pages, but found nothing regarding dtmf tone regeneration, just the indication that inbound tones are not allowed over low bitrate codecs. Would you raccomend sip info or rfc2833 as tone handling method ? P.S. finalmente un compatriota :) MB * hint : did you searched the ml first? MB this has been discussed a lot, even little time ago... MB however... MB sure, just use oob dtmf like rfc2833 or sip info dtmf... MB so you can use a low bitrate codec and asterisk MB will generate them again when going to the pstn... MB matteo MB Il mer, 2004-04-14 alle 10:49, Alessio Focardi ha scritto: Hi, I would like to have some remote users with sip phones over adsl connections access our asterisk pbx and make out calls, currently we are using a zaptel pri interface for outdialing. What is the right way to manage dtmf over pstn lines and still retain low bandwith occupation ? In other words: if I use g729 (and sip info dtmf) for sip phones - asterisk communication will asterisk be able to regenerate real tones when going out to the pstn ? Tnx for any help ... currently I havent got g729 licenses so I cant test it out by myself. -- Matteo Brancaleoni Espia System Administrator Email : [EMAIL PROTECTED] Web : http://www.espia.it Phone : +39 02 70633354 - ext 201 IAX(2): [EMAIL PROTECTED] - ext 201 Iaxtel: 1-700-56-62458 - ext 201 SIP : [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] * Announcement * Astricon 2004 - call for speakers!
eh, very good idea... but how about for alaw people? Any plans to make another conference in EU world? Matteo. P.S. unfortunately I cannot join... too much money for me. Il gio, 2004-04-15 alle 15:16, Olle E. Johansson ha scritto: We're proud to announce Astricon 2004 - the first Asterisk user's and developer's conference! * Where? Atlanta, USA * When? September 22-24, 2004 The conference is arranged in partnership with Digium.inc and the keynote speaker is Mark Spencer, lead developer of Asterisk - the Open Source PBX. Among the speakers already signed on are Ed Guy of Pulver.com, John Todd, Jeremy McNamara (NuFone) and collegues from the SIP Foundry Open Source project. Main topics: * Integrating the PBX with the IT infrastructure: Asterisk for the Enterprise * VOIP migration in-a-box: Asterisk for Service providers * Lower cost, more flexibility: Asterisk for Call Centers * Your VoIP Swiss Army Knife: Asterisk for developers * Managing your Asterisk PBX: from the CLI to the GUI Agenda in brief: * Wednesday: Tutorials - in depth sessions held by VoIP and Asterisk gurus Tutorials will be arranged both for newbies and pro's * Thursday: Conference and exhibition * Friday: Asterisk developer's meeting Early bird registration will start soon at discounted rates on the web site, http://www.astricon.net We're now in the process of setting up the agenda and are looking for speakers and sponsors. Send a tutorial or speaker's proposal to [EMAIL PROTECTED] including * A subject * A brief description (five-six lines) * Target group (if tutorial) * Name and contact information * A digital picture of yourself (for the conference web) We need proposals no later than april 30, 2004. You may of course also propose other speakers than yourself :-) If you're working for a company that sells Asterisk-related products and services, there's an oppurtunity to show your products and sponsor the event. Contact us at [EMAIL PROTECTED] for more information. Looking forward to meeting you all in Atlanta! Steven SokolOlle E. Johansson [EMAIL PROTECTED] [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Matteo Brancaleoni Espia System Administrator Email : [EMAIL PROTECTED] Web : http://www.espia.it Phone : +39 02 70633354 - ext 201 IAX(2): [EMAIL PROTECTED] - ext 201 Iaxtel: 1-700-56-62458 - ext 201 SIP : [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] dtmf for public telephony access
* hint : did you searched the ml first? this has been discussed a lot, even little time ago... however... sure, just use oob dtmf like rfc2833 or sip info dtmf... so you can use a low bitrate codec and asterisk will generate them again when going to the pstn... matteo Il mer, 2004-04-14 alle 10:49, Alessio Focardi ha scritto: Hi, I would like to have some remote users with sip phones over adsl connections access our asterisk pbx and make out calls, currently we are using a zaptel pri interface for outdialing. What is the right way to manage dtmf over pstn lines and still retain low bandwith occupation ? In other words: if I use g729 (and sip info dtmf) for sip phones - asterisk communication will asterisk be able to regenerate real tones when going out to the pstn ? Tnx for any help ... currently I havent got g729 licenses so I cant test it out by myself. -- Matteo Brancaleoni Espia System Administrator Email : [EMAIL PROTECTED] Web : http://www.espia.it Phone : +39 02 70633354 - ext 201 IAX(2): [EMAIL PROTECTED] - ext 201 Iaxtel: 1-700-56-62458 - ext 201 SIP : [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] res_motv: Request for Comment
Hi a) The idea itself -- is it a good one or is it stupid? great idea. could be very useful if you don't have much time to track/test cvs version and/or the bugtracker b) The way to make it deployed without sneaking a call home in on anybody that doesn't want it? make it off by default, providing infos on how to enable it. In this way you don't have to worry about user complaints about privacy (hey, you've turned on! isn't that by default), Also not all systems could have a open internet connection... so sending infos is impossible at all. Matteo. -- Matteo Brancaleoni Espia System Administrator Email : [EMAIL PROTECTED] Web : http://www.espia.it Phone : +39 02 70633354 - ext 201 IAX(2): [EMAIL PROTECTED] - ext 201 Iaxtel: 1-700-56-62458 - ext 201 SIP : [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] res_motv: Request for Comment
Hi. another (stupid) thing. don't call that function motv. motv is a name for another opensource project. Matteo. -- Matteo Brancaleoni Espia System Administrator Email : [EMAIL PROTECTED] Web : http://www.espia.it Phone : +39 02 70633354 - ext 201 IAX(2): [EMAIL PROTECTED] - ext 201 Iaxtel: 1-700-56-62458 - ext 201 SIP : [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Astersik and Europe
Hi Hi There, I got a question is it been possible to use asterisk as an normal telephone... asterisk is a server, not a telephone... unless you wanna fit it into a cpu powered phone :) So far in germany it is not been normal to call a number and then enter the extention number her you dial the number directly... this is called DID , aka direct inward dialing... sure it works with asterisk Matteo. -- Matteo Brancaleoni Espia System Administrator Email : [EMAIL PROTECTED] Web : http://www.espia.it Phone : +39 02 70633354 - ext 201 IAX(2): [EMAIL PROTECTED] - ext 201 Iaxtel: 1-700-56-62458 - ext 201 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Compiling snom firmware
Hi all does anybody here played with the compilation of the snom (100/200) firmware? I'm finding some problems here, but after being able to compile the kernel with the ppc-linux dev kit, create a zvmlinux.initrd image (containing also the filesystem) and creating a .bin file using the romtools... I get always CRC error, and can't load the firmware into the phone (via tftp o web interface) Any hint? Matteo -- Matteo Brancaleoni Espia System Administrator Email : [EMAIL PROTECTED] Web : http://www.espia.it Phone : +39 02 70633354 - ext 201 IAX(2): [EMAIL PROTECTED] - ext 201 Iaxtel: 1-700-56-62458 - ext 201 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Need to interface to BRIs
so you need North America ISDN, not EuroIsdn. The only way is a diva server card with capi driver, Klaus zapBri doesn't support NI (as far as I know) Matteo. Il lun, 2004-02-16 alle 10:53, Jim Archer ha scritto: I forgot to mention, I am in North America. --On Monday, February 16, 2004 4:10 AM -0500 Jim Archer [EMAIL PROTECTED] wrote: Hi All... I would like to interface 4 BRI lines to Asterisk. I looked at Digium's hardware list and, although they have solutions for PRI and T1, I didn't see anything for BRI. I would like to avoid ISDN4Linux if possible. Does anyone know of any hardware suppoted by Asterisk I can use for this? Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Matteo Brancaleoni Espia System Administrator Email : [EMAIL PROTECTED] Web : http://www.espia.it Phone : +39 02 70633354 - ext 201 IAX(2): [EMAIL PROTECTED] - ext 201 Iaxtel: 1-700-56-62458 - ext 201 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Zhone + call transfer
ah ah what is the trasnfer hook time for you phone? Usa phones seems to have a very long time... like 1secs, and that's the default for zaptel... here in italy we have flash time about 120 msec, instead so, try that: edit zaptel.h in your zaptel src dir, search for ZT_DEFAULT_RXFLASHTIME (line 802 in current cvs), and lower the value from 1250 to 200, for example AND lower ZT_MAXPULSETIME (line 805) from (150 * 8) to something like (20 * 8)... but not lower than ZT_MINPULSETIME (15 * 8). compile, install, reload modules, restart asterisk, and let us know. Matteo. Il lun, 2004-02-16 alle 12:50, Kent Williams ha scritto: After finding a spot to put the Zhone Zplex so that the fan noise doesn't annoy anyone, I've got everything working to an acceptable level except for call transfers. No matter what I do the Zhone doesn't seem to be passing 'flash' key presses on to asterisk, ie whenever I try to transfer a call, nothing happens. The DTMF tones pressed after the 'flash' key are simply heard over the conversation. Running asterisk with -vvvc doesn't show anything when trying to transfer a call which leads me to believe that it has something to do with the Zhone. Can anyone confirm that call transfers do in fact work with the Zhone Zplex? Is there anything obvious that I may have missed? ...and yes, I've added the following to Zapata.conf for the appropriate channels: threewaycalling = yes transfer = yes cancallforward = yes Cheers, Kent ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Matteo Brancaleoni Espia System Administrator Email : [EMAIL PROTECTED] Web : http://www.espia.it Phone : +39 02 70633354 - ext 201 IAX(2): [EMAIL PROTECTED] - ext 201 Iaxtel: 1-700-56-62458 - ext 201 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Need to interface to BRIs
Il lun, 2004-02-16 alle 12:49, Cees de Groot ha scritto: Klaus-Peter Junghanns [EMAIL PROTECTED] said: we have a 4 BRI solution for Asterisk, the quadBRI PCI ISDN. One thing I'd like to know about this card: Echo Cancellation? I've replaced by Fritz!Card PCI by a Diva Server 2M, and the difference is remarkable... since is zaptel based, it shares same zaptel routines for EC, as far as I know. Matteo. -- Matteo Brancaleoni Espia System Administrator Email : [EMAIL PROTECTED] Web : http://www.espia.it Phone : +39 02 70633354 - ext 201 IAX(2): [EMAIL PROTECTED] - ext 201 Iaxtel: 1-700-56-62458 - ext 201 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Music on Hold - Context
hi I have set up a * box supporting 3 different companies but have some questions regarding MOH. Can MOH support multiple context or classes. Reason I ask each company would like to have different MOH sound files. Is this possible? yes, just specify multiple moh classes in musiconhold.conf and use each moh class for each company. example: company1 = mp3:/var/lib/asterisk/somemoh1 company2 = mp3:/var/lib/asterisk/somemoh2 company3 = mp3:/var/lib/asterisk/somemoh3 and now assign each moh class on your users/ivr/channels... matteo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Multiple switch staments
Hi. Does anybody ever had the need to use multiple switch staments in one context? like N slave asterisk servers, switching to one master which has in one context N switches to the slaves. so the master only holds a switching table. Any idea? (I know that can be done with a proper dialplan without switches, but making asterisk browse between multiple sw can be useful) matteo. -- Matteo Brancaleoni Espia System Administrator Email : [EMAIL PROTECTED] Web : http://www.espia.it Phone : +39 02 70633354 - ext 201 IAX(2): [EMAIL PROTECTED] - ext 201 Iaxtel: 1-700-56-62458 - ext 201 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] GS and NAT
hi. I've gs working under NAT, simply put nat=yes into sip.conf section if *, then enable nat into the gs, without any stun server. Matteo. Il mar, 2004-02-03 alle 21:17, Tomas Prybil ha scritto: Hi all. Is it at all possible to have a GS B101 NATed with firmware 1.0.4.40? I've tried both STUN and not STUN. The odds seems best with stun because the phone registers with right ip adress. When the connection is made * sends rtp packets to the right destination AND port, but the phone doesn't accept the packets. Should I burn my D-LINK 604 or upgrade the GS? /t ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Matteo Brancaleoni Espia System Administrator Email : [EMAIL PROTECTED] Web : http://www.espia.it Phone : +39 02 70633354 - ext 201 IAX(2): [EMAIL PROTECTED] - ext 201 Iaxtel: 1-700-56-62458 - ext 201 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Playing announcement to called user prior to Confirmation
show application dial from asterisk cli: snip 't' -- allow the called user transfer the calling user 'T' -- to allow the calling user to transfer the call. 'r' -- indicate ringing to the calling party, pass no audio until answered. 'm' -- provide hold music to the calling party until answered. 'H' -- allow caller to hang up by hitting *. 'C' -- reset call detail record for this call. 'P[(x)]' -- privacy mode, using 'x' as database if provided. 'g' -- goes on in context if the destination channel hangs up 'A(x)' -- play an announcement to the called party, using x as file see last param ... Matteo. -- Matteo Brancaleoni Espia System Administrator Email : [EMAIL PROTECTED] Web : http://www.espia.it Phone : +39 02 70633354 - ext 201 IAX(2): [EMAIL PROTECTED] - ext 201 Iaxtel: 1-700-56-62458 - ext 201 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] busy tones
go with early b3 matteo. Il mar, 2004-02-03 alle 12:44, Matteo Rancilio ha scritto: Hi When I call a phone with CAPI if the phone available I hear ringing ok but if the phone is busy I don't hear anything at all. Also, when I call a mobile phone and it is turned off I don't hear the operator voice answer me telling me that the request phone is turned off or unavailable. Any ideas? m ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Matteo Brancaleoni Espia System Administrator Email : [EMAIL PROTECTED] Web : http://www.espia.it Phone : +39 02 70633354 - ext 201 IAX(2): [EMAIL PROTECTED] - ext 201 Iaxtel: 1-700-56-62458 - ext 201 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Mediatrix 1102 Auth
Hi all. I'm evaluating a mediatrix 2fxs 1102. seems great (it has also supervised transfer, that's very needed in office environments and works well). the only I thing I cannot make work is the auth to my asterisk server. If I don't set a password into the mediatrix and *, I can call out, but still the registration goes wrong. using a password, nothing works. I've done some trace with ethereal, comparing the registration process of one sip phone and of the mediatrix. A sip phone registration normally works this way: * phone tries to register * asterisk sends out trying and then a proxy auth required * the phone answers back with the logon data. now the phone is registered. the mediatrix stops at step2: never answers to asterisk with logon data after a proxy auth required any hint? Matteo. -- Matteo Brancaleoni Espia System Administrator Email : [EMAIL PROTECTED] Web : http://www.espia.it Phone : +39 02 70633354 - ext 201 IAX(2): [EMAIL PROTECTED] - ext 201 Iaxtel: 1-700-56-62458 - ext 201 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Choppy Problem!!
Hi i'm managing a call center with asterisk, GS 102 and diva server 4 bri. i have big problem with big choppy sound, In the direction External user --- Agent after a quick phone call with Cristian, we managed to find out 2 things : * hypertreading was enabled and that caused irq errors * capi.conf was wrong Matteo. -- Matteo Brancaleoni Espia System Administrator Email : [EMAIL PROTECTED] Web : http://www.espia.it Phone : +39 02 70633354 - ext 201 IAX(2): [EMAIL PROTECTED] - ext 201 Iaxtel: 1-700-56-62458 - ext 201 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Mediatrix 1102 Auth
Hi. any hint? I've never played with the 1104, however others have reported that it does register correctly when properly configured (and with * properly matching). In order for anyone to offer any suggestions, however, you'll have to pass along the config info for both * and the 1104. Would suggest the sip.conf entry (section) for one extension, and the relavent associated entries for that extension programmed in the 1104. (no passwords please) I managed to make it work. I simply wrote the wrong real into the meadiatrix, since I wrote the * ip addr, instead of asterisk. reverting that, made it register without issues. Matteo. -- Matteo Brancaleoni Espia System Administrator Email : [EMAIL PROTECTED] Web : http://www.espia.it Phone : +39 02 70633354 - ext 201 IAX(2): [EMAIL PROTECTED] - ext 201 Iaxtel: 1-700-56-62458 - ext 201 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Gsm + snom phones
Hi. I'm not using snom phones for a while, but now I want to test again them and I'm gonna buy a snom 200 105 . Some times ago I had a snom 100 , and gsm wasn't working with *. How's now the situation? the snom gsm works well with * ? Thanks for any info, Matteo. -- Matteo Brancaleoni Espia System Administrator Email : [EMAIL PROTECTED] Web : http://www.espia.it Phone : +39 02 70633354 - ext 201 IAX(2): [EMAIL PROTECTED] - ext 201 Iaxtel: 1-700-56-62458 - ext 201 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Gsm + snom phones
Hi. About a month ago I made a test with snom200b. At least then it worked ok with *. At the moment I'm using mainly g711a. So, there is always a possibility something but you also tested gsm ? Greets,Matteo. -- Matteo Brancaleoni Espia System Administrator Email : [EMAIL PROTECTED] Web : http://www.espia.it Phone : +39 02 70633354 - ext 201 IAX(2): [EMAIL PROTECTED] - ext 201 Iaxtel: 1-700-56-62458 - ext 201 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Power Over Ethernet for *any* ethernet switch (or hub); product idea
Hi. The POEI simply connects the four ethernet signals on each of its inputs (pins 1,2,3,6 on each) to the same pins on its corresponding outputs. Additionally, it supplies -48VDC (maybe selectable if there are other voltage needs) on the appropriate pins (also maybe selectable if different vendors use different wiring conventions for POE) of its outputs. and probably you're going to fry something on your lan. POE isn't simple power on the right pins, but is a sort of protocol. Really, on POE enabled devices (or injectors) you won't measure the DC with a tester, simply because POE on port X is enabled after a request by the device on that port. this is for mantaining compatibity with non POE devices. so you will need also something that detects the power request on each port and enables it. Matteo. -- Matteo Brancaleoni Espia System Administrator Email : [EMAIL PROTECTED] Web : http://www.espia.it Phone : +39 02 70633354 - ext 201 IAX(2): [EMAIL PROTECTED] - ext 201 Iaxtel: 1-700-56-62458 - ext 201 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CAPI not installed, after changed from i4l to CAPI
: unused variable `error' chan_capi.c:207: warning: unused variable `CMSG' chan_capi.c:208: warning: unused variable `buf' chan_capi.c: In function `capi_send_digit': chan_capi.c:253: warning: unused variable `error' chan_capi.c:254: warning: unused variable `CMSG' chan_capi.c:255: warning: unused variable `buf' gcc -shared -Xlinker -x -o chan_capi.so chan_capi.o -lcapi20 gcc -pipe -Wall -Wmissing-prototypes -Wmissing-declarations -g -I/usr/include/asterisk -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DNEVER_EVER_EARLY_B3_CONNECTS -DFORCE_SOFTWARE_DTMF -DCAPI_ULAW -DCAPI_GAIN -DDEFLECT_ON_CIRCUITBUSY -DCAPI_SYNC -Wno-missing-prototypes -Wno-missing-declarations -DCRYPTO -c -o app_capiCD.o app_capiCD.c gcc -shared -Xlinker -x -o app_capiCD.so app_capiCD.o gcc -pipe -Wall -Wmissing-prototypes -Wmissing-declarations -g -I/usr/include/asterisk -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DNEVER_EVER_EARLY_B3_CONNECTS -DFORCE_SOFTWARE_DTMF -DCAPI_ULAW -DCAPI_GAIN -DDEFLECT_ON_CIRCUITBUSY -DCAPI_SYNC -Wno-missing-prototypes -Wno-missing-declarations -DCRYPTO -c -o app_capiHOLD.o app_capiHOLD.c gcc -shared -Xlinker -x -o app_capiHOLD.so app_capiHOLD.o gcc -pipe -Wall -Wmissing-prototypes -Wmissing-declarations -g -I/usr/include/asterisk -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DNEVER_EVER_EARLY_B3_CONNECTS -DFORCE_SOFTWARE_DTMF -DCAPI_ULAW -DCAPI_GAIN -DDEFLECT_ON_CIRCUITBUSY -DCAPI_SYNC -Wno-missing-prototypes -Wno-missing-declarations -DCRYPTO -c -o app_capiRETRIEVE.o app_capiRETRIEVE.c gcc -shared -Xlinker -x -o app_capiRETRIEVE.so app_capiRETRIEVE.o gcc -pipe -Wall -Wmissing-prototypes -Wmissing-declarations -g -I/usr/include/asterisk -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DNEVER_EVER_EARLY_B3_CONNECTS -DFORCE_SOFTWARE_DTMF -DCAPI_ULAW -DCAPI_GAIN -DDEFLECT_ON_CIRCUITBUSY -DCAPI_SYNC -Wno-missing-prototypes -Wno-missing-declarations -DCRYPTO -c -o app_capiECT.o app_capiECT.c gcc -shared -Xlinker -x -o app_capiECT.so app_capiECT.o gcc -pipe -Wall -Wmissing-prototypes -Wmissing-declarations -g -I/usr/include/asterisk -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DNEVER_EVER_EARLY_B3_CONNECTS -DFORCE_SOFTWARE_DTMF -DCAPI_ULAW -DCAPI_GAIN -DDEFLECT_ON_CIRCUITBUSY -DCAPI_SYNC -Wno-missing-prototypes -Wno-missing-declarations -DCRYPTO -c -o app_capiMCID.o app_capiMCID.c gcc -shared -Xlinker -x -o app_capiMCID.so app_capiMCID.o rm app_capiCD.o app_capiECT.o app_capiMCID.o app_capiHOLD.o app_capiRETRIEVE.o Any suggestions.. and sorry for a very long posting... -- Matteo Brancaleoni Espia System Administrator Email : [EMAIL PROTECTED] Web : http://www.espia.it Phone : +39 02 70633354 - ext 201 IAX(2): [EMAIL PROTECTED] - ext 201 Iaxtel: 1-700-56-62458 - ext 201 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] fw: problem with safe_asterisk
have you changed the line where is ask to use another console (default tty9) ? Il mar, 2004-01-13 alle 23:07, Pat Boyle ha scritto: I checked the log files in /var/log/asterisk There was nothing in there related to these errors. I think the script is ending after before asterisk even starts. Pat --- Karsten Wemheuer [EMAIL PROTECTED] Tue, 13 Jan 2004 08:34:31 +0100 a.. Previous message: [Asterisk-Users] Fw: problem with safe_asterisk b.. Next message: [Asterisk-Users] MeetMe issues? c.. Messages sorted by: [ date ] [ thread ] [ subject ] [ author ] Hi, Pat Boyle wrote: I have no problems lauching asterisk from the command line . . . asterisk -c However, I'm trying to autostart on boot up, so I'm trying safe_asterisk When I do this, I get: Asterisk ended with exit status 127. Asterisk died with code 127. Aborting. I've looked through the script but can't find what the problem is. I'm running on RedHat Fedora. Could You please have a look in the logfile. Maybe there are some information about the abort. I don't use Fedora but on Debian the log is under /var/log/asterisk/messages HTH, Karsten ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Matteo Brancaleoni Espia System Administrator Email : [EMAIL PROTECTED] Web : http://www.espia.it Phone : +39 02 70633354 - ext 201 IAX(2): [EMAIL PROTECTED] - ext 201 Iaxtel: 1-700-56-62458 - ext 201 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Why I can not use the conference
meetme requires zaptel
Il mer, 2004-01-14 alle 08:56, Zhang Peihao ha scritto:
Hi All,
The meetme.conf have created as below:
[rooms]
conf = 101
conf = 102
and extensions.conf as below:
exten = _1XX,1,MeetMe,${EXTEN}
why the warning printed when I called 101.
WARNING[27660]: File pbx.c, Line 1051 (pbx_extension_helper): No application
'MeetMe' for extension (ipcentrex, 101, 1)
And I found asterisk have not load the meetme.conf when it starts up.
Zhang Peihao
[EMAIL PROTECTED]
2004-01-14
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Matteo Brancaleoni
Espia System Administrator
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Re: [Asterisk-Users] Best Linux Distribution
the one you feel most confortable with. as far as I know, asterisk is developed under RedHat, but really, I run it with RH, debian, slack. Many with suse and so on... so is up to you. matteo. Il mar, 2004-01-13 alle 12:48, [EMAIL PROTECTED] ha scritto: Hi my question is: which is the best distribution to work with asterisk? thanks mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Matteo Brancaleoni Espia System Administrator Email : [EMAIL PROTECTED] Web : http://www.espia.it Phone : +39 02 70633354 - ext 201 IAX(2): [EMAIL PROTECTED] - ext 201 Iaxtel: 1-700-56-62458 - ext 201 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Best Linux Distribution
cool idea :) Il mar, 2004-01-13 alle 13:10, Daniel Bichara ha scritto: [EMAIL PROTECTED] wrote: Hi my question is: which is the best distribution to work with asterisk? Hi Mark, I am working on a distro called SAX built to optimize * and routing. It works with RPMs and its HFS is RedHat like. I built all packages by hand and created RPMs packages. It is in beta version by now. More few days and I will release an ISO image. Daniel thanks mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Matteo Brancaleoni Espia System Administrator Email : [EMAIL PROTECTED] Web : http://www.espia.it Phone : +39 02 70633354 - ext 201 IAX(2): [EMAIL PROTECTED] - ext 201 Iaxtel: 1-700-56-62458 - ext 201 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Nufone.net net wackiness?
only domain name screwed up. mmh.. my registrar allows me an autorenew for all domain names... pretty useful :) matteo. Il mar, 2004-01-13 alle 09:24, Chris Albertson ha scritto: Looks like they went off the air just after my PayPal payment was processed. I gues we wait a couple days to see if Nufone has gone belly up/bankrupt/gone or if this is just a domain name screw up. --- Steven Critchfield [EMAIL PROTECTED] wrote: On Tue, 2004-01-13 at 01:26, Brian Capouch wrote: I can't send mail to any addresses in nufone.net; they all get rejected by a spam blocker. And their website is gone, too!! The URL leads to a parking site. = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Yahoo! Hotjobs: Enter the Signing Bonus Sweepstakes http://hotjobs.sweepstakes.yahoo.com/signingbonus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Matteo Brancaleoni Espia System Administrator Email : [EMAIL PROTECTED] Web : http://www.espia.it Phone : +39 02 70633354 - ext 201 IAX(2): [EMAIL PROTECTED] - ext 201 Iaxtel: 1-700-56-62458 - ext 201 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] pick up remote call
is just *8 see ya. matteo. Il mar, 2004-01-13 alle 16:03, massimo ha scritto: Hi, I,m trying to pickup remote call using the SIP protocol and *8# from my phone but with no success. I just installed * 0.7.0 and my Phones are connected to one ATA 186 with image 2.16.1. I set in the sip.conf the follow parameter: callgroup=1 pickupgroup=1 for each phone. Someone can help me ? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Matteo Brancaleoni Espia System Administrator Email : [EMAIL PROTECTED] Web : http://www.espia.it Phone : +39 02 70633354 - ext 201 IAX(2): [EMAIL PROTECTED] - ext 201 Iaxtel: 1-700-56-62458 - ext 201 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] E100P - connected to Cisco
08 08 02 80 00 4e 18 03 a9 83 9f 79 01 80 [02 01 06 08 08 02 80 00 4e 18 03 a9 83 9f 79 01 80 ] [02 01 06 08 08 02 80 00 4e 18 03 a9 83 9f 79 01 80 ] Informational frame: SAPI: 00 C/R: 1 EA: 0 TEI: 000EA: 1 N(S): 003 0: 0 N(R): 004 P: 0 13 bytes of data -- ACKing all packets from 3 to (but not including) 4 -- Since there was nothing left, stopping T200 counter -- Stopping T203 counter since we got an ACK -- Nothing left, starting T203 counter Protocol Discriminator: Q.931 (8) len=13 Call Ref: len= 2 (reference 32768/0x8000) (Terminator) Message type: RESTART ACKNOWLEDGE (78) Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 31 ] Restart Indentifier: [ Ext: 1 Spare: 0 Resetting Indicated Channel (0) ] Sending Receiver Ready (4) [ [02 [02 01 [02 01 01 [02 01 01 08 [02 01 01 08 ] [02 01 01 08 ] Supervisory frame: SAPI: 00 C/R: 1 EA: 0 TEI: 000EA: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] N(R): 004 P/F: 0 0 bytes of data -- Restarting T203 counter -- Restarting T203 counter Thanks, Daniel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Matteo Brancaleoni Espia System Administrator Email : [EMAIL PROTECTED] Web : http://www.espia.it Phone : +39 02 70633354 - ext 201 IAX(2): [EMAIL PROTECTED] - ext 201 Iaxtel: 1-700-56-62458 - ext 201 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Mailing list growth
I joined the ML on july 12 2002, and my Evolution Folder has 20486 msg on Asterisk-Users, 2149 on Asterisk-dev and 182 on Asterisk-doc :) keep up the good work! matteo. Il gio, 2004-01-08 alle 09:48, Olle E. Johansson ha scritto: So far in January, we've had 726 messages on -users. December 2003: 2.978 messages November 2003: 3.410 messages October 2003: 3.177 messages December 2002: 741 messages December 2001: 67 messages ...the project is growing. /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Matteo Brancaleoni Espia System Administrator Email : [EMAIL PROTECTED] Web : http://www.espia.it Phone : +39 02 70633354 - ext 201 IAX(2): [EMAIL PROTECTED] - ext 201 Iaxtel: 1-700-56-62458 - ext 201 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
