Re: [Asterisk-Users] Segregating a test version of asterisk - libpri/zaptel locations
Julien Levi wrote: I'm beginning to think a chroot environment may be the best route - * fails to compile as it loads the wrong headers from libpri and zaptel (those from the may 2004 install, the new ones get put in /usr/local/new/usr/include). I got it to compile by editing the makefile INCLUDE line to: INCLUDE=-I/usr/local/new/usr/include -Iinclude -I../include However * fails to load with this error: /usr/local/new/usr/lib/asterisk/modules/chan_zap.so: undefined symbol: pri_sr_set_channel I set LD_LIBRARY_PATH to /usr/local/new/usr/lib:/usr/local/new/lib but I think the above error is due to chan_zap still trying to use the old libpri. I'm going to keep experimenting in the hope of getting this working without using chroot. If I do I'll write it up on the wiki for other clueless neophytes ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Segregating a test version of asterisk - libpri/zaptel locations
Tzafrir Cohen wrote: install them under prefix /usr/local/new and add /usr/local/new/lib to the beginning of LD_LIBRARY_PATH . Also consider using a idfferent machine or a chroot environment for testing. A chroot environment is not as complicated as it sounds. Thanks for the tip. I'm beginning to think a chroot environment may be the best route - * fails to compile as it loads the wrong headers from libpri and zaptel (those from the may 2004 install, the new ones get put in /usr/local/new/usr/include). I've never set one up before though, so wish me luck! I can't use a separate machine as we only have one Linux server and telephony card. I suppose I could just overwrite lipri and zaptel and only install * to a different location but that defeats the point of segregating the install which is not to touch the working installation. ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Segregating a test version of asterisk - libpri/zaptel locations
I currently run an asterisk server with cvs from May 2004. I'm planning to upgrade to the latest stable version but want to segregate a test version first. I know I can do this by editing the install_prefix field in the makefile. I can also change the install prefix and load the new zaptel module manually for testing. However what is the situation then with libpri? If I Install this to another location, how do I tell asterisk to look there for it? Is the only option to overwrite the version of libpri in /usr/lib and copy back the old version if things don't work? Thanks in advance for any help. ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Bluetooth with *
Martin List-Petersen wrote: Check http://www.crazygreek.co.uk/content/chan_bluetooth, but it's still in heavy development. Far from finished. Isn't there also a module to allow location tracking via bluetooth, that is, the room you are in is triangulated via bluetooth and your calls are routed to the nearest phone? I'm sure I remember reading something about it at one point... ___ 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] GPL Violations (Was: Advice on OS Choice)
Jason Becker wrote: I also think GPL violations are rare. But there was a highly publicized alleged violation by Cisco/Linksys: http://lkml.org/lkml/2003/6/7/164 To be honest I don't even know the end of that story (if it has ended)... probably some bureaucratic snafu. AFAIK Linksys admitted their error and published the source, including the full build environment. http://www.linksys.com/support/gpl.asp This has resulted in a number of open source firmware projects that add features to the router in question and other Linksys appliances that use Linux. Linksys/Cisco are now seen as one of the better companies with respect to the gpl. ___ 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] New MOH stream for each queue member?
Hi there, Is it possible to start a new MOH stream for each queue member? My queue MOH is a message rather than music and I want customers to enter the queue and hear it from the start rather than from some random point. The first person entering the queue always seems to hear it from the start. I know could (ab)use the position announcement settings (which interrupts the MOH), changing the message and disabling the actual position announcement, but my understanding of how this works is that it would block queue members from exiting the queue whilst the announcement was playing (which I don't want). Any tips on getting this working - would it require changes to the code? I'm not a programmer. Thanks in advance for any help. regards, Julien ___ 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] BT101 and caller id and web interface
Simon wrote: 2. When i get an incoming call * ( BT ) strips the leading 0 from the callerid. Is it possible to put this back on ? would i use setcallerid ? Have you tried putting: nationalprefix=0 internationalprefix=00 In your zapata.conf file? This should help if you are using isdn. I don't know about analog though. regards, Julien ___ 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] Multiple DDI Hunting on Analog Lines (UK)
Matt wrote: Hi everyone, I want to get multiple DDI's and hunting across those DDI's in case one of the lines is busy using analog phone lines. The system is for a large house so I want 3 x PSTN lines. 3 x DDI's and the ability for those DDI's to be presented across all three PSTN lines. BT say you can't have more than one DDI number associated with a PSTN line and that you can't hunt across analog lines. snip Line hunting across analog lines is possible and it is free. BT call it Auxiliary working. Take a look at this page: http://www.aaisp.net.uk/aa/twolines.html It's not something they often do on home lines so be prepared for sales staff who don't know what they are talking about. DDI on analog lines is more complicated and I believe there is a set-up charge of over £1000 (see http://www.serviceview.bt.com/list/current/docs/Exch_Lines.boo/1228.htm ) If you must have multiple numbers you'd be better off with an isdn line. regards, -- Julien ___ 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] Multiple DDI Hunting on Analog Lines (UK)
Julien Levi wrote: DDI on analog lines is more complicated and I believe there is a set-up charge of over £1000 (see http://www.serviceview.bt.com/list/current/docs/Exch_Lines.boo/1228.htm ) If you must have multiple numbers you'd be better off with an isdn line. To emulate a couple of DDI numbers, it may be possible to get distinctive ring service (It's the Call Sign select service in BT speak) and route calls depending on the ring pattern. I don't know if this is possible with Auxiliary working lines though... regards, -- Julien ___ 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] Odd behaviour with asterisk -rx
Hello, I was planning to use the output of asterisk -rx show queues in a script when I noticed that sometimes asterisk only outputs the first line of the response. e.g: debian:/# asterisk -rx zap show channels Chan Extension Context Language MusicOnHold debian:/# asterisk -rx zap show channels Chan Extension Context Language MusicOnHold debian:/# asterisk -rx zap show channels Chan Extension Context Language MusicOnHold 1incoming 2incoming 4incoming 5incoming debian:/# asterisk -rx show queues bookingsqhas 1 calls (max 100) in 'ringall' strategy (9s holdtime), C:28, A:3, SL:57.1% within 20s debian:/# asterisk -rx show queues bookingsqhas 1 calls (max 100) in 'ringall' strategy (9s holdtime), C:28, A:3, SL:57.1% within 20s Members: SIP/agent2 (dynamic) has taken 11 calls (last was 386 secs ago) SIP/agent1 (dynamic) has taken 16 calls (last was 1566 secs ago) Callers: 1. Zap/4-1 (wait: 0:08) debian:/# I'm on asterisk 1.0_stable - has this been fixed in head, is it a known issue? I was unable to find anything about it in a search of previous list posts... regards, Julien Levi ___ 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 TOS - worrying for an open source project?
Andrew Kohlsmith wrote: You will note it said NONexclusive ... right -- they're not saying the information belongs to them, they are saying that the information is in the public domain. Yes, the right is non-exclusive but they claim copyright over the compilation (see the copyright statement in the attached file) AIUI that means that should anyone ever want to mirror the wiki elsewhere they would need to either have a license from voip-info.org or seek individual permission from everyone who ever contributed, which would be practically impossible. A creative commons license would avoid such issues. What worries me most is that the current terms seem crafted so as to ensure that should the people who run voip-info ever decide to remove content, or stop hosting the wiki, it couldn't be mirrored anywhere else. Untrue. Their terms about relinking or republishing are for COMMERCIAL use, unless I'm misreading something here. As I read it each restriction is separated by a comma, therefore the reproduce, duplicate, copy restrictions are separate from the sell, resell, visit or use for other commercial purposes restrictions. The full text of the TOS from: http://www.voip-info.org/terms_of_service.html is attached. regards, Julien Terms of Service Welcome to voip-info.org. Arte Marketing Inc, (hereafter referred to as voip-info.org) and its affiliates provide their services to you subject to the following conditions. If you visit or use our website, you accept these conditions. Please read them carefully. PRIVACY Please review our Privacy Policy, which also governs your visit to voip-info.org. COPYRIGHT All content included on this site, including but not limited to: text, graphics, logos, button icons, images, audio clips, digital downloads, data compilations, and software, is the property of voip-info.org or its suppliers and protected by United States and international copyright laws. The compilation of all content on this site is the exclusive property of voip-info.org and protected by U.S. and international copyright laws. All software used on this site is the property of voip-info.org or its suppliers and protected by United States and international copyright laws. TRADEMARKS voip-info.org and other marks indicated on our site are registered trademarks of voip-info.org. voip-info.org graphics, logos, page headers, button icons, scripts, and service names are trademarks or trade dress of voip-info.org. or its subsidiaries. voip-info.org's trademarks and trade dress may not be used in connection with any product or service that is not voip-info.org's, in any manner that is likely to cause confusion among customers, or in any manner that disparages or discredits voip-info.org. All other trademarks not owned by voip-info.org or its subsidiaries are the property of their respective owners. LICENSE AND SITE ACCESS voip-info.org grants you a limited license to access and make personal use of this site. This license does not include any resale or commercial use of this site or its contents. Without express written consent of voip-info.org you may not: Download (other than page caching), or modify this site. Reproduce, duplicate, copy, sell, resell, visit or use for other commercial purposes this site or any portion thereof. Use frames or framing techniques to enclose this site or any portion thereof for commercial purposes. Use meta tags or other 'hidden text' utilizing voip-info.org's name or trademarks. Any unauthorized use terminates the permission or license granted by voip-info.org. You are granted a limited, revocable, and nonexclusive right to create hyperlink(s) to our pages so long as the link does not portray voip-info.org, its affiliates, or their products or services in a false, misleading, derogatory, or otherwise offensive matter. You may not use any voip-info.org logo or other proprietary graphic or trademark as part of the link without express written permission. YOUR ACCOUNT If you create and/or use an account on this site, you are responsible for all activities by this account and agree to keep the account password and other access information confidential. If you are under 18, you may use voip-info.org only with involvement of a parent or guardian. voip-info.org and its affiliates reserve the right to refuse service, terminate accounts, remove or edit content, or cancel orders in their sole discretion. REVIEWS, COMMENTS, COMMUNICATIONS, AND OTHER CONTENT Visitors may enter content into various areas of the voip-info.org website, only if the content is not illegal, obscene, threatening, defamatory, invasive of privacy, infringing of intellectual property rights, or otherwise injurious to third parties and does not contain software viruses, political campaigning, commercial solicitation, chain letters, mass mailings, or any form of spam. You may not misrepresent your identity by using false e-mail address,
Re: [Asterisk-Users] SIP Registration Problem
Brian Rathman wrote: I am using snom200 phones registering with Asterisk via SIP. I can see where the phone registers without a problem, and then when you try and make a call I get a proxy authentication required message on the phone and failed to authenticate user error in the Asterisk messages file. Then the next call you make from the phone goes through without a problem. Nothing changes between these two events, but it is almost like the phone is using two different passwords for the same account. Has anyone else seen a problem like this? I am using an Asterisk CVS version from early March, not sure if upgrading will help as well. Thanks, Brian Please don't start a new thread by replying to an exisiting post - threaded mailreaders list it as a reply to that post (even if you change the subject, as theading is done by messageid). You're also less likely to get a response due to the post being inside an existing converstion rather than as listed as a new topic. regards, Julien ___ 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] app_queue and app_groupcount
Troy Settle wrote: I just disable call waiting on all my sip phones and on all zap interfaces. No problem. That is fine if it can be done, though I prefer to keep as much set-up info on the server as possible for easier admin. Do you know of a softphone with such an option? I've been unable to find one that rejects calls when on line is busy. regards, -- drbob ___ 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] app_queue and app_groupcount
The new app_groupcount looks great for most applications but it a is a step back for call queueing... since app_queue calls physical interfaces and not extensions, app_groupcont can't be used to limit the calls passed to a dynamically added agent. I presently use the broken sip incominglimit feature (even though it's less than ideal as it also limits outgoing calls preventing consultative transfer using sip refer commands) I could start to use the agents app with agentcallbacklogin to (almost) emulate the current behaviour and use app_groupcount - I can automate the login using agentcallback login, but not the logoff, it prompts for an extension to forward to requireing # to pressed to log off - is there any way round this? I'd prefer to keep the simplicity of simply dialing one number to log on in or out of the queue from any phone, without having to define agentids, passwords, etc which we don't need. I hope incominglimit and outgoing limit aren't going to be removed entirely... -- Julien ___ 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] Problems with Quadbri card
Hello, We have been running a system based around the quadbri card from www.junghanns.net for around 3 weeks now. For the first two weeks everything was stable and ran well. In the last week a issue has appeared, described below: Someone attempts to call us, * sees an incoming call (it is anounced on the console e.g. Accepting call from '' to '781950' on channel 2, span 2) and * says it has picked up the call. However the person calling isn't connected, hears no ringing and gets a tone as if the call has been rejected. This happens on some but not all calls. As * believes the call is connected the channel remains open and the call remains in the system until it is manually cleared. It doesn't appear to be a service provider issue, they;ve tested the lines and see no problem. The only strange error message in the logs is: WARNING[15376]: PRI: !! Don't know what to do with M3=7 u-frames which has started to appear recently bu it occurs once to twice a day where as the failed calls can happen tens of times a day. We have 2 isdn2e line with 1 hunt number. We were on bristuff 0.0.2rc20a and have changed to the latest 0.0.2 but the problem still occurs. Any idea on how to resolve this would be greatly appreciated. regards, Julien Levi ___ 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 Quadbri card
Also, since installing 0.0.2 we see this occasionally in the logs: May 19 18:59:26 WARNING[16400]: Ring requested on channel 1 already in use on span 1. Hanging up owner. May 19 18:59:32 WARNING[16400]: Ring requested on channel 2 already in use on span 1. Hanging up owner. May 19 19:00:10 WARNING[16400]: Call specified, but not found? May 19 19:00:26 WARNING[16400]: Call specified, but not found? ___ 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] CallCenter setup
Maciek Kaminski wrote: Hi, I am investigating possibility of using asterisk as an call center controller, i.e. Clients phone in, interact with IVR, if IVR is not enough get redirected to human consultant. There should be possibility for supervisors to connect to ongoing conversation. Expected traffic will not exceed 30 concurrent calls. Asterisk box should be connected to Siemens communication platform HiPath 3750 that controls whole telephony system. This Siemens has ISDN and VOIP(H323 as I have been told) interfaces. Now my problem is which interface to choose? Will voip be good enough? Wont it introduce to much latency? Or should I insist on buying ISDN interface for asterisk box? What hardware would You recommend for this setup? VOIP will always give more latency than than an isdn interface. However we have an entirely voip based system (for internal calls) and the latency isn't noticeable. Another question: anyone has successfully deployed call center solution using soft phones? If yes, with which soft phones? Our (very small, 3 agents) call centre uses all softphones. We currently use x-lite but are looking at iaxcomm to intergrate the phone with the CRM software. I'm happy with the setup but have been having some intermittent faults with incoming calls recently that I believe may be a problem with the quadbri card we use. Hope this helps - anyone out there with a larger setup willing to comment? regards, Julien ___ 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] i4l -- capi move - how?
Mark Elkins wrote: I have * with i4l installed and working - on a dumb eicon card. It seems in order to get DTMF out of the BRI (for business banking - etc) - I should change from i4l drivers to capi drivers. wiki help seems to be for the Fritz card only...??? I have ticked the suggested boxes in 'menuconfig', explored the capi4linux websites - etc... but am missing some magic. I've modprobe capi, if I don't modprobe hisax type=11 to get- HiSax: Card 1 Protocol EDSS1 Id=HiSax (0) HiSax: Eicon.Diehl Diva driver Rev. 1.1.4.2 ... then whet am I meant to do instead? Obviously no * (load_module: CAPI not installed!) A pointer in the right direction please? ie - does the Dumb EICON card work with capi?, do I need the server eicon card instead?, is CAPI only working with Fritz's cards? As far as I know the only passive ISDN cards with capi drivers are the AVM fritz range. You can't at present use chan_capi with any other card. If the eicon card is based on the HFC chipset you may be able to use the zaphfc drivers and address it as a native zaptel device. Take a look at http://www.junghanns.net/asterisk/ for bristuff Otherwise consider buying a cheap second hand fritz (AKA BT speedway card) or hfc chipset card from ebay. There are usually quite a few available. regards, -- Julien ___ 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] time of waiting in queues
Daniel Cubero Salas, Ing. wrote: We have implement a IVR with CTD+CDR and queues, all on asterisk. Just for statistics, we need to save the time which one client was waiting in queue. Someone knows asterisk has a function than it can be load in a module (programming by us) or if a module with this function was developtment. Thanks in advance 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 What version of asterisk are you running? I believe the latest cvs version has a queue logging feature which should report this information. -- Julien ___ 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] Analogue telephone cards for the UK
Kevin Walsh wrote: The nice people at TelAppliant will sell you an analogue FXO card, and are based in London, England. See here: http://www.voiptalk.org/ The Digium X100P (well, the X101P now) works in England with the notable exception of support for BT's caller ID. Some UK cable companies (eg NTL or Telewest) use bellcore (US) caller id in certain areas but they use BT standard in others. The only way to be certain to get caller id with * at the moment is to use an ISDN line (this will require an ISDN line card, not the x101p). The new FXO (external line) ports (available soon) for the TDM400P will be _capable_ of receiving BT's caller id but whether support for it gets added into the driver is a different matter. regards Julien ___ 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] quadBRI card installation issues
Robinson Tim-W10277 wrote: Use RC16. This seems to solve our issues on a UK ISDN2e line. Rgds Tim Thanks for the tip - rc16 didn't work but rc17 magically fixed things. I also had to ensure that the dip switches on my by highway box were set to S and IN regards, Julien ___ 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] quadBRI card installation issues
Hi there, I am attempting to set up a simple BRI and SIP based platform using * with the quadbri card (it's not sharing an IRQ). I enclose my zaptel and zapata.conf files. For the inital test I'm simply trying to connect to the * demo menu. The drivers compile (with a few warning that I believe aren't important - see attachments). chan_zap comiles with the warning: chan_zap.c: In function `pri_dchannel': chan_zap.c:6344: warning: passing arg 1 of `pbx_builtin_setvar_helper' from incompatible pointer type The qozap driver appears to load correctly and I get this in the log : Apr 1 17:51:07 debian kernel: Zapata Telephony Interface Registered on major 196 Apr 1 17:51:07 debian kernel: qozap: start Apr 1 17:51:07 debian kernel: PCI: Enabling device 00:0b.0 ( - 0003) Apr 1 17:51:07 debian kernel: PCI: Found IRQ 10 for device 00:0b.0 Apr 1 17:51:07 debian kernel: qozap: quadBRI card configured at mem 0xd0888000 IRQ 10 HZ 100 CardID 0 Apr 1 17:51:07 debian kernel: S/T port 1 is in TE mode. Apr 1 17:51:07 debian kernel: S/T port 2 is in TE mode. Apr 1 17:51:07 debian kernel: S/T port 3 is in TE mode. Apr 1 17:51:07 debian kernel: S/T port 4 is in TE mode. Apr 1 17:51:07 debian kernel: qozap: registered zaptel span 1. Apr 1 17:51:07 debian kernel: qozap: registered zaptel span 2. Apr 1 17:51:07 debian kernel: qozap: registered zaptel span 3. Apr 1 17:51:07 debian kernel: qozap: registered zaptel span 4. Apr 1 17:51:07 debian kernel: card 1 span 1 state F0 (A_ST_RD_STA = 0x0) Apr 1 17:51:07 debian kernel: card 1 span 2 state F0 (A_ST_RD_STA = 0x0) Apr 1 17:51:07 debian kernel: card 1 span 3 state F0 (A_ST_RD_STA = 0x0) Apr 1 17:51:07 debian kernel: card 1 span 4 state F0 (A_ST_RD_STA = 0x0) Apr 1 17:51:07 debian kernel: qoztmp-cardno = 1 Apr 1 17:51:07 debian kernel: qozap: 1 multiBRI card(s) in this box, 4 BRI ports total. Apr 1 17:51:07 debian kernel: Registered tone zone 4 (United Kingdom) Apr 1 17:51:07 debian kernel: qozap: starting card 1 span 1/0. Apr 1 17:51:07 debian kernel: card 1 span 1 state F6 (A_ST_RD_STA = 0x16) Apr 1 17:51:07 debian kernel: card 1 span 1 state F7 (A_ST_RD_STA = 0x17) However when running * I get the message below every 2-3 seconds: Apr 1 18:10:55 WARNING[11276]: PRI: !! Got S-frame while link down Attempting to call the line does not result in it being answered but I get the error: Apr 1 18:11:23 WARNING[11276]: PRI: !! Got I-frame while link state 0 When the line starts to ring and again when I hang up. I'm using a bt buisness highway line which is isdn2e comaptible but doesn't provide power on the digital socket. Any suggestions on how to resolve this would be greatly appreciated. I can find nothing on this in the list archives (though similar errors have been seen using a t410p card under high call load: http://lists.digium.com/pipermail/asterisk-users/2004-March/040745.html ) I'm using the bri-stuff rc15 from: http://www.junghanns.net/asterisk/downloads/bri-stuff-0.0.2rc15.tar.gz Thanks in advance for any suggestions, regards, Julien cc -Wall -Wstrict-prototypes -Wmissing-prototypes -g -c -o pri.o pri.c pri.c: In function `pri_hangup': pri.c:253: warning: implicit declaration of function `q921_hangup' cc -Wall -Wstrict-prototypes -Wmissing-prototypes -g -c -o q921.o q921.c q921.c:134: warning: no previous prototype for `q921_send_teireq' q921.c:302: warning: no previous prototype for `q921_send_sabme' q921.c: In function `__q921_receive': q921.c:1227: warning: array subscript has type `char' q921.c:1228: warning: array subscript has type `char' q921.c: At top level: q921.c:235: warning: `q921_send_teiverify' defined but not used cc -Wall -Wstrict-prototypes -Wmissing-prototypes -g -c -o prisched.o prisched.c cc -Wall -Wstrict-prototypes -Wmissing-prototypes -g -c -o q931.o q931.c q931.c: In function `transmit_calling_party_number': q931.c:895: warning: control reaches end of non-void function q931.c: In function `q921_getcall': q931.c:1423: warning: assignment from incompatible pointer type q931.c:1432: warning: assignment from incompatible pointer type q931.c: At top level: q931.c:1776: warning: no previous prototype for `q931_information_special' q931.c:1836: warning: no previous prototype for `q931_hold_acknowledge' q931.c:1843: warning: no previous prototype for `q931_hold_reject' q931.c:2107: warning: no previous prototype for `q921_hangup' q931.c: In function `q921_hangup': q931.c:2133: warning: assignment from incompatible pointer type q931.c: In function `q931_receive': q931.c:2256: warning: unused variable `ttei' q931.c:2255: warning: unused variable `i' q931.c: At top level: q931.c:1284: warning: `transmit_my_paging_signal' defined but not used ar rcs libpri.a pri.o q921.o prisched.o q931.o ranlib libpri.a cc -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes -g -o pri.lo -c pri.c pri.c: In function `pri_hangup': pri.c:253: warning: implicit declaration of function `q921_hangup' cc -fPIC
[Asterisk-Users] Few questions from new user...
Hi there, I've just recently discovered Asterisk via the website and it looks perfect for what I want to do. I've read the handbook draft but still have some questions: I work for a very small company and have almost no budget for hardware. Ideally we'd like to have 3-4 analogue FXO lines coming into the server (a p3 600, 512mb ram which we already own) via 4 pci voice modems. From there we'd use Asterisk to provide automatic call distribution to up to 3 operators (most of time there will only be 1 to 2 operators) using software ip phones on their pc's (preferably with on hold music and a message saying what position in the queue people are while they wait). Is such a setup possible, and can anybody recommend the model of hardware voice modem which I should use? I've found these intel modems which look like they may fit the bill but I don't know if they have linux drivers: http://www.upgradecenterinc.com/inclipin56kv.html Also I can see in the handbook how to script different routings during open and closed hours (where a we are closed message is played out of hours) but I was wondering if it is possible to have a manual switch between open and closed - sometimes we stay open for 30 minutes extra if we're very busy and having the phone system just shut down at 3am would be too inflexible. Sorry for the long post I've more questions but I'll save them for another time. I realise all this has probably been covered in the past but I've found navigating the list archives pretty difficult with no search option available. Thanks in advance for any help. Regards, -- Julien ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
