[Asterisk-Users] Using locked PAP2 and PAP2-NA with Asterisk
Here is some info that may allow some "locked" PAP2 and PAP2-NA units to be used with Asterisk: I have a PAP2-NA (from a provider other than Vonage) for which I did not know the admin password, though the "user" pages were accessible to me. The provider had set it up to fetch at startup, its configuration file by HTTP from a numeric IP. It was running 2.0.10(LSc). A search of the wiki found the Sipura mass deployment page, which says "NOTE: Recent versions of the SPA-2000 and SPA-3000 firmware ( > 2.0.11), and the Linksys PAP2, also support the plaintext XML configuration format." That suggested a solution in two steps: upgrade the firmware, then feed in an XML file. PAP2-bin-2-00-13-LSb.bin is readily available on the Net; after making it accessible to my TFTP server, a visit to http:///upgrade?tftp:/// caused the PAP-NA to upgrade successfully. Then, a PC on an isolated LAN was configured with the IP from which the PAP2-NA was trying to read its config. The PC was set to provide the following response: HTTP/1.0 200 OK Content-Type: text/xml 123456 foo Setting the domain is not part of the hack, but it makes it easy to see that it worked. You refresh the user status page, if you see the domain change to foo, you can then go to the admin page and login with 123456. Emboldened by the easy solution, I went to a local store, bought a PAP2 (sold for use with Vonage), and hooked it up on an isolated LAN. First, I tried "reset" from the IVR menu. Strangely, it was accepted without a password and the unit rebooted, but nothing got reset. Next, since the unit had 2.0.9(LSd), I tried to upgrade. Shoot, it asked for the admin password. However, I decided to continue anyway. The unit does a DNS lookup for ls.tftp.vonage.net, so I set it to use the PC as DNS server, which was suitably configured to provide the PC's IP as the answer. Next, the unit is trying to to TFTP to port 2400. My TFTP server wasn't port agile, so I found one that was. But then, it's back on port 69, asking for /spa.xml . I put the xml file (starting with 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 duration limits not working
Yeah, I have CVS-HEAD as of today and it's still no good. Thoughts? > I have CVS-HEAD as of yesterday and it's still not working for me. Maybe > I'll try updating again and post the results. > > Thanks, > Tim > >> [EMAIL PROTECTED] wrote: >>> Hello everybody. Recently I've been trying to limit the duration of >>> some >>> calls for a simple application I'm writing. Unfortunately all of the >>> documented methods are failing and I'm not sure what else to try. Here >>> are some samples of what I've done: >> >> I believe this was fixed in CVS-HEAD a couple of weeks ago. There was a >> big announcement on the mailing list about 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 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
Re: [Asterisk-Users] hybrid clients
On 8/21/05, Scott Huang <[EMAIL PROTECTED]> wrote: > Is it possible to connect a IAX softphone to a SIP softphone via Asterisk? > > IAX client -- Asterisk -- SIP client > > I tried that, and I was able to dial and talk to my IAX client from the SIP > client. But not the other way around, I couldn't dial the SIP client from > the IAX client. The SIP client was not ringing. Asterisk showed some WARNING > messages: It is compleetly possible, it sounds like the problem you have could be due to nat traversal on the sip client - sip does not like nat traversal. Also make sure both phones are registering using "sip show peers" and "iax show peers" from the asterisk cli. -- Thanks, Lance Grover ___ 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] hybrid clients
Hi, folks, Is it possible to connect a IAX softphone to a SIP softphone via Asterisk? IAX client -- Asterisk -- SIP client I tried that, and I was able to dial and talk to my IAX client from the SIP client. But not the other way around, I couldn't dial the SIP client from the IAX client. The SIP client was not ringing. Asterisk showed some WARNING messages: Aug 21 20:24:38 WARNING[11044]: chan_sip.c:1047 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Critical Request) == No one is available to answer at this time (1:0/0/0) What was wrong? Your help is highly appreciated. -Scott ___ 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] IP Cop as a firewall and QOS
I can sign that immediately. I am not using asterix yet, but I am having VoIP phone behind IpCop and never had a problem yet. About the SOHO design you mention, the only limitation it has is that you can only have a single green network (internally subnet), but you can abuse the blue (designed for WLAN clients) and the orange(DMZ) for that too. And I think for version 1.5 it is planned to have multiple green network possible. But I think you should go to the IpCop users newsgroup and ask there if it suits your special needs and if somebody already has a config like yours. Austin Denyer wrote: On Wed, 2005-08-17 at 17:27 -0500, Mojo Jojo wrote: I don't mind buying an appliance to get something solid but IP Cop just looks better than he appliances I see out there. I am only concerned if it is stable for a production environment. It says it's designed for a SOHO environment, we are doing a bit more than that. Will this thing hold up? Can it be trusted? I'm not using IPCop with * (I'm very much a * newbie), but I am using it as a general firewall, and it rocks. I have had no issues with it, and I have been running IPCop for several years. It is very stable - I have yet to have it crash on me. It is secure - the box has yet to be successfully hacked (and the logs show numerous attempts on a daily basis!) It will handle your bandwidth easily as long as your hardware is not too antiquated. For example, I've got it running on a 133MHz Pentium, 128Mb RAM on a 3MB/sec connection, and it hardly even notices... Try it - you'll like it. Regards, Austin. ___ 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] Suggestions
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Please price me on this. I need a service provider that does this. Like the guy that mentioned collocation. Could this be done and for how much? Joshua 1) What type of phones do you plan to use (analog, SIP, Skinny, H.323, MGCP)? SIP 2) How many phones will you operate? 3 Phones 3) How many external lines do you have, and of which type (analog, BRI, PRI, T1, VoIP)? 1 Analog Cable -- VOIP 4) How many concurrent internal/external calls do you expect (example: 30%)? Consult the Erlang tables (see links below) if in doubt. 40%+ 5) What codecs will be employed, and will you need to do a lot of transcoding? Tip: Enter SHOW TRANSLATION at the CLI. SUGGEST - NEED TO RECORD ALL INCOMING/OUTGOING CALLS AND HAVE VM 6) What features shall the system provide (echo cancellation, voicemail, conferencing queues & call center, recording, fax, voice menu, text-to-speech, speech recognition)? Automatic Call Delivery Echo Cancellation Voicemail Fax to Email Conferencing Queues Call Center Recording of incoming, outgoing FAX Voice Menus Text-to-speech Speech Recognition Password Protection before outgoing call Call Forwarding Call Blocking Call Parking Call Transfer Caller ID Caller ID Block Privacy Manager Music On Hold Calling Card Setup 6 Second Billing Wake-up Call Scheduler (On-line Calendar) Do Not Disturb Page/Intercom Distinctive Rings International Dialing 911 Service 7) How reliable/scalable must your system be? Must be vary realiable (POSED 2 Servers, 1 backup) 8) How many Asterisk boxes do you plan to put in place? See #7 9) What does your IP network look like (speed, QoS support, VLAN, Power-over-Ethernet)? 1.544Mbps - 2.000Mbps I would like to setup QoS Support Other Notes: Vonage (Or some other easier VOIP provider in Missouri) GossipTel FWD IPTel IConnect SIPPhone Freenumber IPStar.us V911.us CallVantage myvoipline PLUS MUCH MORE (If you could come up with a one stop solution for the above providers that would be great) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.1 (GNU/Linux) iD8DBQFDCUqSeYokiwGEZc4RAmpbAKCu70f6k8CVkR/es5tTmr57Bpf7FQCfTpkl g591sQEtDa9SW6PLqargdfI= =NEcE -END PGP SIGNATURE- begin:vcard fn:Joshua Abbott (Successful Hosting) n:Abbott;Joshua org:Successful Hosting;Support adr:3009 Avenue J;;Attn: Joshua Abbott;Brooklyn;NY;11210;USA email;internet:[EMAIL PROTECTED] title:Technical Support Representative tel;work:+1 (866) 494-5096 x1207 tel;fax:+1 (419) 858-3241 note:Alt E-Fax: (801) 217-1123 x-mozilla-html:FALSE url:http://www.successfulhosting.com/ version:2.1 end:vcard ___ 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] Unable to create channel of type 'Zap'
My zaptel.conf config: -
# Below setting is for E1
span=1,1,0,cas,hdb3
bchan=1-15
bchan=17-31
dchan=16
loadzone = us
defaultzone=us
My zapata.conf config: -
# Below setting is for E1
switchtype = national
signalling = pri_cpe
group = 1
channel => 1-15
channel => 17-31
My extension.conf config: -
[default]
exten => 181,1,Dial(Zap/1/181)
When I perform a dailing from my SIP Phone, I got the
error message as below: -
-- Executing Dial("SIP/118301-6f4e",
"Zap/1/181") in new stack
Aug 22 11:03:02 NOTICE[9023]: app_dial.c:764
dial_exec: Unable to create channel of type 'Zap'
== Everyone is busy/congested at this time
I am beginner...How to solve this?
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[Asterisk-Users] Re: call waiting beep on PSTN and TDM400P FXO linehook flash
I had a similar issue with sending a flash to the PSTN for call waiting. I found that my dial plan in the extensions.conf file was not allowing me to dial *xx. Once I corrected my dial plan I was able to dial *0, *69, *78, *79, etc. Training the wife on how to actually use it was an entirely different issue. I have not tried to enable 3-way calling via the PSTN. "Jeff Otterson" <[EMAIL PROTECTED]> wrote in message news:<[EMAIL PROTECTED]>... > I have been looking for the answer to this question for a > while. Google-ing and reading the archives of Asterisk-Users has not > enlightened me. > >It seems that this question has been asked many times, and many times it > has gone unanswered. > > I have call waiting and three way calling on my PSTN line from Verizon > (the local telco). This is connected to a FXO port on a TDM400P. I also > have two FXS ports on the TDM400P. > >So my problem is, how do I flash the Verizon PSTN line when I hear the > call waiting beep? How can I send a hook flash to the Verizon trunk to > activate their 3-way calling feature. > >I have seen some stuff like "hook flash then send *0 to get the bridged > Zap trunk to flash" but I can't get it to work. I get the reorder > signal. I need something my wife and kid can do. > >Can anybody help? > >Thanks, > >Jeff > > ___ > 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
Re: [Asterisk-Users] TDM11B modprobe wcfxs fails
> > I got my TDM11B and am trying to get it to work on my PC. > > However, I'm having difficulty getting the wcfxs driver to > > load. I've Googled this problem, and while there are others ... > 1. The TDM card has several different revisions (rev e through h, > I believe). If you have one of the later revisions, you may need > zaptel software later then stable v1.07. Thanks! It turned out to be this case. My card was a rev I and needed 1.0.9 or later. I now have my card working and so is Asterisk! Thanks again! Fumi Okushi ___ 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] All Page ??
Steve Maroney wrote: Does anyone know of any plans to add an intercom/all-page feature in *? The few SIP phones that have auto-answer capability would be better if Asterisk could broadcast one leg of a channel to many legs at one time. I'm looking for an answer to this problem also. I am putting an Asterisk system into our new office. In our old office we used the old phone system to act as an intercom, you hit page all and your voice comes out of the speaker on several handsets throughout the office. This allows you to announce information or to the whole office, simply announcing to someones desk doesn't work since our people move around a lot and are not always at their desk. Anyway, I have some Polycom phones, and I have Autoanswer working with Asterisk, but which ever phone happens to answer the call first is the only one who's speaker my voice comes out of. Anyone have an answer to this problem? Thanks, Matt ___ 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] perl-cpan
Dear list, I was installing Asterisk via the AMP method off the AMP website. There is a portion in there where they want you to use perl-cpan to install telnet. The first time I installed I had no problem. I messed up and trashed the box further down in the install.. This time I made a mistake and put a -jp3 switch for multi processors. This has caused the install of telnet tank. I found the cpan config file and removed it. This allowed me to get past the first mistake but now when I try to complie telnet it fails and I do not get any error's on the make file. How can I remove perl-cpan and start again, or am I facing another trip to the colo to splash the box and start again? Thank you for your time, Tommy ___ 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] Vtech/Vonage Wireless Phone system
I currently have one of the Hitachi WIP-5000 SIP phones. I've been using it off and on for about four months. Just one as an initial test. While it's pretty good it does have some minor issue, or they could be issues related to my wlan access points. Whatever the case, it drops off lan periodically causing a silent period in the current call. I've had to extend the qualify setting to 3000 or it gets reported as not available now and again. My next test will be the Aastra/Sayson 480 cti SIP deskphone with cordless handset. It costs about the same as the Hitachi phone on its own. I don't think that the cordless handset is truly SIP/Wifi based. But that may not matter if its well integrated into the phone. Michael On Sun, 21 Aug 2005 18:47:50 -0400, Andre Normandin wrote: >I already have a Sipura and a panasonic cordless, I was just hoping that I >could find a true SIP wireless phone at a reasonable price :-) > >Oh well, one can drea > >Thanks, > - Andre > >-Original Message- >From: [EMAIL PROTECTED] >[mailto:[EMAIL PROTECTED] Behalf Of Michael >Graves >Sent: Sunday, August 21, 2005 4:10 PM >To: Asterisk Users Mailing List - Non-Commercial Discussion >Subject: RE: [Asterisk-Users] Vtech/Vonage Wireless Phone system > > >All these things are is just an ATA built into a plain vanilla analog >cordless phone system...ok, a 5.8 Ghz systemvanilla with sprinkles >;-) > >Nothing special at all. You'd be better off with a Sipura device and a >Panasonic cordless phone. > >Michael > >On Sun, 21 Aug 2005 15:12:49 -0400, Andre Normandin wrote: > >>Thanks, >> >>I just read all the literature on the Vtech website, and, I think you are >>exactly correct! >> >>Oh well, so much for cheap SIP wireless phones :-) >> >>-Original Message- >>From: [EMAIL PROTECTED] >>[mailto:[EMAIL PROTECTED] Behalf Of Rich >>Adamson >>Sent: Sunday, August 21, 2005 2:51 PM >>To: Asterisk Users Mailing List - Non-Commercial Discussion >>Subject: Re: [Asterisk-Users] Vtech/Vonage Wireless Phone system >> >> >>> I just stopped in at Best Buy here in CT, USA. I found an interesting >>> offering from Vtech there. It states it's a VOIP wireless phone system >>made >>> for the Vonage service. >>> >>> Here it is at their website: >>> http://www.vtechphones.com/vtechui/guide/dsp_voip_guide.cfm >>> >>> For $149, it comes with 2 (supposidly) VOIP 802.11b wireless phones, and >a >>> 'broadband/Wireless' router. >>> >>> Does anyone know anything about these? Are they SIP phones? If so, I >>wonder >>> if it's possible to modify them for use with Asterisk? Do they need >their >>> 'base' to talk, or is that nothing more than an 802.11b wireless router? >>> >>> I'm thinking of picking up a kit just to see if I can get them to work >>> nativly with Asterisk, but if anyone has any experience with them before >I >>> do, I'd appreciate it.. >> >>I seen their ad in the paper this morning. It appeared the phones were >>the standard cordless (not 802.11b), and the base unit had pc wireless >>(802.11b) integrated with the linksys/sipura broadband stuff. You might >>want to check that out a little closer; but, fairly good return policy >>if the ads are a little misleading. ;) >> >> >>___ >>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 >> >> > >-- >Michael Graves [EMAIL PROTECTED] >Sr. Product Specialist www.pixelpower.com >Pixel Power Inc. [EMAIL PROTECTED] > >o713-861-4005 >o800-905-6412 >c713-201-1262 >fwd 54245 > > > >___ >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 > > -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 fwd 54245 ___ 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/list
[Asterisk-Users] SDP media attribute
Hello, Any use of SDP media attribute in conjunction with SIP /Asterisk ? I would appreciate any insight! George ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ 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] Dial Zero to get outside line?
Hello, My asterisk currently will dial an outside number after I dial the number and press send on the phone. How can I get it setup so I have to press '0' for an outside line. Kind regards Michael Felder IT Medic Australia Pty. Ltd. P: 03 9557 2213 F: 03 9557 2214 M: 0419 568 217 E: [EMAIL PROTECTED] http://www.ITMedic.com.au Keeping your computer systems healthy. ___ 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] Call duration limits not working
I have CVS-HEAD as of yesterday and it's still not working for me. Maybe I'll try updating again and post the results. Thanks, Tim > [EMAIL PROTECTED] wrote: >> Hello everybody. Recently I've been trying to limit the duration of >> some >> calls for a simple application I'm writing. Unfortunately all of the >> documented methods are failing and I'm not sure what else to try. Here >> are some samples of what I've done: > > I believe this was fixed in CVS-HEAD a couple of weeks ago. There was a > big announcement on the mailing list about 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 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] dtmf tones during conversation
Help for this use case: I need detect a dtmf tone during conversation. For example: A call B B answer A A and B talk B (while talk) press *111 and I defined an action en extensions.conf to exten => *111,1,Application() and while the application is executed A and B continues talk --- What application would have to use? Thanks and regards Leonardo ___ 1GB gratis, Antivirus y Antispam Correo Yahoo!, el mejor correo web del mundo http://correo.yahoo.com.ar ___ 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] Voice Problem ...asterisk on lan not woriking well...!!
Does your linksys support traffic shaping in any way? If so you should mark your VoIP data as a high priority. Alternatively, use a slimer codec like G729. I'm not sure what the Polycom supports but It'll probably do 729. You'll have to buy some licences for your * box too. Mark Ing. Marlo R. Beltran G wrote: Hi i just implemented asterisk and is such a grate solution...i am using polycom 301 and 501 phoneson lan a iam using g.711 and i have a 16 port linksys switch... the problem come when somebody inside the network is making a call to other extension (in the same network) and is sending an e mail through internet the quality goes down...it hears rally bad... i am on a 10/100 network with cat5e on wire, and switches...what can i do to have an excelent voice quality inside my network??? The e mail doesn't go trough the asterisk computer. Marlo ___ 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 -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com ___ 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] Small office setup/using analog lines w/ Asterisk
Thanks, everyone, for your suggestions. I'm going to stop by the office tomorrow to try some of these out. Here's more info on the setup: We bought a brand new computer for this -- I don't have the specifics right now, but will look that up in the office tomorrow. We have two Digium cards -- a TDM04B and a TDM01B. This system supports two organizations sharing the same office space: 3 lines go to one org., the other 2 go to the other. The Digium cards actually are sharing IRQs with other devices -- the installer mentioned it could be an issue initially, but when he saw that the devices that the cards were sharing with were the network card and the video card, he said to just try and see if it works first. Sounds like some of the problems that we're having could be related to this, and it's probably the first thing I should try changing. I have tried using ztmonitor to adjust the txgain and rxgain. It was very frustrating ... changing the tx to higher or lower values did not necessarily change it in the direction you'd think. Also, changing rx sometimes affected tx and vice versa. In the end, I gave up because nothing I did increased sound quality -- although it wasn't so hard to make things worse. Any hints here? We're using Sipura phones. We realize these aren't has high quality as Polycoms, but this was what the manger decided to get based on the budget. We're spending more than originally anticipated now because of the problems we're having, but it seems like the Digium cards are a more likley culprit than the Sipura phones. However, if there's a way to isolate problems, that'd be great. I guess one thing is we could try more experiments with one extension calling another -- if the sound quality is a lot better, then it's unlikely a problem with the Sipura phones. If the sound quality is poor, I'll try hooking up the phones to a new network card. As for interference ... we don't have wireless devices, and the main phone we're testing with is about 10' away from the switch. I suppose there's still some possibility with interference, but I think it's something I'll prioritize a little lower. One additional question -- are VoIP lines generally easier to get going w/ good sound quality than POTS lines? One reason we went with POTS was for sound quality. Of course, we also figured it'd be more reliable than DSL, which was the main reason for going with POTS lines. Right now we use 5 POTS lines ... if VoIP sounds better, we were thinking of dropping three of those lines and getting VoIP lines, keeping the POTS lines as backup. Any thoughts on that plan? One thing I'm not sure about is how to select the POTS lines from the phone sets. I suppose we could assign a second extension to the phones (they support 2) and have the second one be POTS and the first be VoIP. Another thing I was wondering is whether we could get hunting to work properly with a mix of VoIP and POTS. I'll call the phone company tomorrow, but if anyone has tried anything like this, I'd like to hear about it. Also, are there any VoIP vendors that work particularly well with Asterisk? I've seen Broadvoice mentioned a bit ... are they a good company for this? I'd love a company that could do automatic failover -- if the VoIP line cuts out, transfer to an analog phone line (I know Vonage has a feature like this, but they're expensive and we don't need a lot of the features). Thanks again for all the suggestions! Jen ___ 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] CVS-HEAD Compile Problem
Nico, Same problem for me, did you find a way to compile latest CVS-HEAD ? And by the way, you're right realtime do need HEAD version. Thanks Laurent At 00:04 20/08/2005 +0200, Nico Giefing wrote: but the non head version is not working with realtime configuration? hm, i think its a problem with app_expr.c but i will try now to copy the app_expr.c from cvs-version i will let you know Nico - Original Message - From: "Trey Scarborough" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <[EMAIL PROTECTED]> Sent: Friday, August 19, 2005 8:31 PM Subject: Re: [Asterisk-Users] CVS-HEAD Compile Problem > I ran into the same problem the other day and just went back to non head > version It would be nice to figure out why it does this. > > - Original Message - > From: Nico Giefing > To: Asterisk Users Mailing List - Non-Commercial Discussion > Sent: Friday, August 19, 2005 9:20 AM > Subject: [Asterisk-Users] CVS-HEAD Compile Problem > > > I have a little Problem, > > I will compile asterisk CVS-HEAD but after 20 second of compiling i get the > message as shown at http://pastebin.com/340654 about 1000 times. > > Do anybody know a solution for this? > > Thanks a lot > > Nico > > > > ___ > 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
[Asterisk-Users] Voice Problem ...asterisk on lan not woriking well...!!
Hi i just implemented asterisk and is such a grate solution...i am using polycom 301 and 501 phoneson lan a iam using g.711 and i have a 16 port linksys switch... the problem come when somebody inside the network is making a call to other extension (in the same network) and is sending an e mail through internet the quality goes down...it hears rally bad... i am on a 10/100 network with cat5e on wire, and switches...what can i do to have an excelent voice quality inside my network??? The e mail doesn't go trough the asterisk computer. Marlo ___ 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] Broadvoice Issue
Mark,
Thanks for the tips. After adding the exten => XX,1,BLAH, i am able to
received calls, however I still get the same error when dialing out, and now,
there is an additional error on the end. I am beginning to think this is a
Broadvoice issue and will try to contact them after sending this message. The
new error is as follows:
===
-- Executing Dial("SIP/202-7ea7", "SIP/[EMAIL PROTECTED]") in new stack
-- Called [EMAIL PROTECTED]
Aug 21 17:33:23 NOTICE[20742]: chan_sip.c:8648 handle_response: Failed to
authenticate on INVITE to '"Cisco 02" ;tag=as6a8b6a73'
== Spawn extension (agents, number, 1) exited non-zero on 'SIP/202-7ea7'
-- Got SIP response 481 "Call Leg/Transaction Does Not Exist" back from
147.135.20.128
===
Can anyone tell from this message if their service is trying to reinvite? I
have that set to no for the devices i'm using as well as for the
[sip.broadvoice.com].
Doing a sip show registry shows me as registered, however, I still cannot make
calls. Any other suggestions?
Thanks,
Josh
-Original Message-
From: [EMAIL PROTECTED] on behalf of Mark Phillips
Sent: Sun 8/21/2005 4:53 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Broadvoice Issue
There seems to be a random thread of BV issues this last week all
amounting to the same proble - no calls.
Do a sip debug peer sip.broadvoice.com and see what happens. I found
that BV were sending calls to my number and for some odd reason my *
server wasn't dumping them into the exten=>s,1,blah logic that it
previously had been. The only way I could fix it was to do
exten=>phonenumber,1,blah and that works fine now.
As for outgoing, are you sure you are registered?
In the shoirt term, log in to your bv account and set up your VM. At
least you won;t lose calls that way.
Mark
Tressler, Joshua Adam wrote:
> I did a quick google search of the lists site and couldn't find a
> definitive answer, so if it's there, I apologize for asking again.
>
>
>
> Starting about noon yesterday, I am no longer able to send/receive calls
> via Broadvoice. When calling in, I get a fast busy, and when calling out
> I get the following error:
>
>
>
> -- Executing Dial("SIP/112-572a",
> "[EMAIL PROTECTED]") in new stack
>
> -- Called [EMAIL PROTECTED]
>
> Aug 21 13:34:47 NOTICE[20742]: chan_sip.c:8648 handle_response: Failed
> to authenticate on INVITE to '"Mobile"
> ;tag=as124e3440'
>
> == Spawn extension (agents, 78126631234, 1) exited non-zero on
> 'SIP/112-572a'
>
>
>
> I have the following in sip.conf:
>
>
>
>
>
> register =>
> [EMAIL PROTECTED]:password:[EMAIL PROTECTED]/
>
>
>
> [sip.broadvoice.com]
>
> type=peer
>
> user=phone
>
> host=sip.broadvoice.com
>
> fromdomain=sip.broadvoice.com
>
> fromuser=XX
>
> secret=password
>
> insecure=very
>
> context=incoming
>
> authname=XX
>
> dtmfmode=inband
>
> dtmf=inband
>
> canreinvite=no
>
>
>
>
>
> Does anyone know what I'm missing here? Everything was working fine
> yesterday morning.
>
>
>
>
>
> JT
>
>
>
>
> ___
> 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
--
Mark, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com
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Re: [Asterisk-Users] Call duration limits not working
[EMAIL PROTECTED] wrote: Hello everybody. Recently I've been trying to limit the duration of some calls for a simple application I'm writing. Unfortunately all of the documented methods are failing and I'm not sure what else to try. Here are some samples of what I've done: I believe this was fixed in CVS-HEAD a couple of weeks ago. There was a big announcement on the mailing list about 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] Vtech/Vonage Wireless Phone system
I already have a Sipura and a panasonic cordless, I was just hoping that I could find a true SIP wireless phone at a reasonable price :-) Oh well, one can drea Thanks, - Andre -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Michael Graves Sent: Sunday, August 21, 2005 4:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Vtech/Vonage Wireless Phone system All these things are is just an ATA built into a plain vanilla analog cordless phone system...ok, a 5.8 Ghz systemvanilla with sprinkles ;-) Nothing special at all. You'd be better off with a Sipura device and a Panasonic cordless phone. Michael On Sun, 21 Aug 2005 15:12:49 -0400, Andre Normandin wrote: >Thanks, > >I just read all the literature on the Vtech website, and, I think you are >exactly correct! > >Oh well, so much for cheap SIP wireless phones :-) > >-Original Message- >From: [EMAIL PROTECTED] >[mailto:[EMAIL PROTECTED] Behalf Of Rich >Adamson >Sent: Sunday, August 21, 2005 2:51 PM >To: Asterisk Users Mailing List - Non-Commercial Discussion >Subject: Re: [Asterisk-Users] Vtech/Vonage Wireless Phone system > > >> I just stopped in at Best Buy here in CT, USA. I found an interesting >> offering from Vtech there. It states it's a VOIP wireless phone system >made >> for the Vonage service. >> >> Here it is at their website: >> http://www.vtechphones.com/vtechui/guide/dsp_voip_guide.cfm >> >> For $149, it comes with 2 (supposidly) VOIP 802.11b wireless phones, and a >> 'broadband/Wireless' router. >> >> Does anyone know anything about these? Are they SIP phones? If so, I >wonder >> if it's possible to modify them for use with Asterisk? Do they need their >> 'base' to talk, or is that nothing more than an 802.11b wireless router? >> >> I'm thinking of picking up a kit just to see if I can get them to work >> nativly with Asterisk, but if anyone has any experience with them before I >> do, I'd appreciate it.. > >I seen their ad in the paper this morning. It appeared the phones were >the standard cordless (not 802.11b), and the base unit had pc wireless >(802.11b) integrated with the linksys/sipura broadband stuff. You might >want to check that out a little closer; but, fairly good return policy >if the ads are a little misleading. ;) > > >___ >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 > > -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 fwd 54245 ___ 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] call waiting beep on PSTN and TDM400P FXO line hook flash
I have been looking for the answer to this question for a while. Google-ing and reading the archives of Asterisk-Users has not enlightened me. It seems that this question has been asked many times, and many times it has gone unanswered. I have call waiting and three way calling on my PSTN line from Verizon (the local telco). This is connected to a FXO port on a TDM400P. I also have two FXS ports on the TDM400P. So my problem is, how do I flash the Verizon PSTN line when I hear the call waiting beep? How can I send a hook flash to the Verizon trunk to activate their 3-way calling feature. I have seen some stuff like "hook flash then send *0 to get the bridged Zap trunk to flash" but I can't get it to work. I get the reorder signal. I need something my wife and kid can do. Can anybody help? Thanks, Jeff ___ 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] Call duration limits not working
Hello everybody. Recently I've been trying to limit the duration of some
calls for a simple application I'm writing. Unfortunately all of the
documented methods are failing and I'm not sure what else to try. Here
are some samples of what I've done:
1) The AbsoluteTimeout application.
- exten => 1,1,AbsoluteTimeout (30)
2) The new version of AbsoluteTimeout.
- exten => 1,1,Set(TIMEOUT(absolute)=30)
3) The S option to the Dial application.
- exten => 1,1,Dial(${PROVIDER}/${NUMBER}|30|S(30))
4) The L option to the Dial application.
- exten => 1,1,Dial(${PROVIDER}/${NUMBER}|30|L(3))
5) A combination of all of the above.
- exten => 1,1,Set(TIMEOUT(absolute)=30) OR exten =>
1,1,AbsoluteTimeout (30)
- exten => 1,2,Dial(${PROVIDER}/${NUMBER}|30|S(30)L(3))
Nothing seems to work. When using "S" I get a message on the console that
says the call is going to be limited to 30 seconds but it just keeps
going. For completeness I'm dialing an IAX provider and I'm using the
Asterisk CVS HEAD. I also am sure that did not forget to reload (and
sometimes restart entirely) after I made changes to extensions.conf. The
timeout "|30" works when calls don't supervise but the time limit "|S(30)"
or "|L(3)" lets calls continue forever when calls do supervise.
What could I be doing wrong? What should I try to remedy this situation?
Thank you,
Tim Mattison
[EMAIL PROTECTED]
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Re: [Asterisk-Users] Broadvoice Issue
There seems to be a random thread of BV issues this last week all
amounting to the same proble - no calls.
Do a sip debug peer sip.broadvoice.com and see what happens. I found
that BV were sending calls to my number and for some odd reason my *
server wasn't dumping them into the exten=>s,1,blah logic that it
previously had been. The only way I could fix it was to do
exten=>phonenumber,1,blah and that works fine now.
As for outgoing, are you sure you are registered?
In the shoirt term, log in to your bv account and set up your VM. At
least you won;t lose calls that way.
Mark
Tressler, Joshua Adam wrote:
I did a quick google search of the lists site and couldn’t find a
definitive answer, so if it’s there, I apologize for asking again.
Starting about noon yesterday, I am no longer able to send/receive calls
via Broadvoice. When calling in, I get a fast busy, and when calling out
I get the following error:
-- Executing Dial("SIP/112-572a",
"[EMAIL PROTECTED]") in new stack
-- Called [EMAIL PROTECTED]
Aug 21 13:34:47 NOTICE[20742]: chan_sip.c:8648 handle_response: Failed
to authenticate on INVITE to '"Mobile"
;tag=as124e3440'
== Spawn extension (agents, 78126631234, 1) exited non-zero on
'SIP/112-572a'
I have the following in sip.conf:
register =>
[EMAIL PROTECTED]:password:[EMAIL PROTECTED]/
[sip.broadvoice.com]
type=peer
user=phone
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
fromuser=XX
secret=password
insecure=very
context=incoming
authname=XX
dtmfmode=inband
dtmf=inband
canreinvite=no
Does anyone know what I’m missing here? Everything was working fine
yesterday morning.
JT
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--
Mark, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com
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[Asterisk-Users] Nortel Meridian-1 Line Side E1
Anyone has the settings to connect a TE405 to Meridian-1 line side E1? I saw T1 on the voip-info.org, but no E1. Is Nortel's E1 a variant of MFC/R2? ___ 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] Vtech/Vonage Wireless Phone system
All these things are is just an ATA built into a plain vanilla analog cordless phone system...ok, a 5.8 Ghz systemvanilla with sprinkles ;-) Nothing special at all. You'd be better off with a Sipura device and a Panasonic cordless phone. Michael On Sun, 21 Aug 2005 15:12:49 -0400, Andre Normandin wrote: >Thanks, > >I just read all the literature on the Vtech website, and, I think you are >exactly correct! > >Oh well, so much for cheap SIP wireless phones :-) > >-Original Message- >From: [EMAIL PROTECTED] >[mailto:[EMAIL PROTECTED] Behalf Of Rich >Adamson >Sent: Sunday, August 21, 2005 2:51 PM >To: Asterisk Users Mailing List - Non-Commercial Discussion >Subject: Re: [Asterisk-Users] Vtech/Vonage Wireless Phone system > > >> I just stopped in at Best Buy here in CT, USA. I found an interesting >> offering from Vtech there. It states it's a VOIP wireless phone system >made >> for the Vonage service. >> >> Here it is at their website: >> http://www.vtechphones.com/vtechui/guide/dsp_voip_guide.cfm >> >> For $149, it comes with 2 (supposidly) VOIP 802.11b wireless phones, and a >> 'broadband/Wireless' router. >> >> Does anyone know anything about these? Are they SIP phones? If so, I >wonder >> if it's possible to modify them for use with Asterisk? Do they need their >> 'base' to talk, or is that nothing more than an 802.11b wireless router? >> >> I'm thinking of picking up a kit just to see if I can get them to work >> nativly with Asterisk, but if anyone has any experience with them before I >> do, I'd appreciate it.. > >I seen their ad in the paper this morning. It appeared the phones were >the standard cordless (not 802.11b), and the base unit had pc wireless >(802.11b) integrated with the linksys/sipura broadband stuff. You might >want to check that out a little closer; but, fairly good return policy >if the ads are a little misleading. ;) > > >___ >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 > > -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 fwd 54245 ___ 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] any ISDN/PRI signaling experts out there?
> -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Paul Belanger > Sent: Friday, August 19, 2005 4:26 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] any ISDN/PRI signaling experts out there? > > See comments inline! > > Damon Estep wrote: > > I have officially engaged in a pissing contest with the local Telco over > > PRI calling name delivery. > > Welcome to my world, I deal with theses guys daily! Errgiant arn't > they. We have a saying around work 'The telco is always wrong!'. > > > The telco publishes their calling name delivery over PRI feature as > > being bellcore gr-1367-core compliant. > > > > The gr-1367-core spec states that the calling name is to be included as > > a facility IE in the setup message, or sent in a subsequent facility IE > > message with an indicator in the setup message that the CNAM will > > follow. > > > > Extensive testing and ISDN/PRI protocol analysis shows that the facility > > IE they are sending out with the CNAM in it comes only after we have > > sent back PROGRESS and ALERTING in response to the SETUP. If we block > > the PROGRESS and ALERTING and sit and WAIT for the FACILITY we never get > > it, the call will time out, so we know they are actually waiting for the > > call to progress before sending the facility IE CNAM. > > This sounds a little fishy, Orgination Number is usually transmitted in > the SETUP message. Your are almost correct in your messaging: > > Network User(Switch) > Setup > CALL PROCEEDING > ALERTING > CALL CONNECT > CALL CONNECT ACKNOWLEDGE > > > There is about a 4sec timeout allow after SETUP is initially sent, if > CALL PROCEEDING is not transmitted by that time, the Network side will > terminiate the call. > > > As far as I can tell the GR-1367-CORE spec does not define a maximum > > delay in sending the facility IE or whether it is acceptable to wait for > > PROGRESS and ALERT before sending it. > > > > The setup is; Telco PRI Lucent 5ESS <> Lucent MAX TNT <> Asterisk > > Here is an ISDN trace from a Dialogic board attached to a 5ESS switch > with framing/coding ESF/B8ZS: > > SETUP(0x05) > 1: BEARER CAPABILITY(0x04) > 2: IE Length(0x03) > 3: 1--- Extension Bit > -00- Coding Standard > ---0 Info. Transfer Cap. > 4: 1--- Extension Bit > -00- Transfer Mode > ---1 Info. Transfer Rate > 5: 1--- Extension Bit > -01- Layer 1 Indent > ---00010 User Info. Layer 1 > 1: CHANNEL ID(0x18) > 2: IE Length(0x03) > 3: 1--- Extension Bit > -0-- Interface ID Present > --1- Interface Type > ---0 Spare > 1--- Preferred/Exclusive > -0-- D-Channel Indicator > --01 Info. Channel Sel. > 3.2: 1--- Extension Bit > -00- Coding Standard > ---0 Number Map > 0011 Channel/Map Element > 4: 1--- Extension Bit > -001 Channel Number/Slot Map > 1: CALLING PARTY NUM(0x6c) > 2: IE Length(0x0b) > 3: 1--- Extension Bit > -010 Type Of Number > 0001 Numbering Plan ID > 949459 Number Digit(s)<-- Here is the ANI > 1: CALLED PARTY NUM(0x70) > 2: IE Length(0x04) > 3: 1--- Extension Bit > -100 Type of Number > 0001 Numbering plan ID > 200 Number Digit(s) <-- Here is the DNIS > > Notice my comments on where ANI and DNIS arrive in the SETUP message. > > > The MAX TNT responds to the Facility IE with ISDN error 98, invalid > > message for call state. > > This is an actual CAUSE CODE from Q.931: > > Cause No. 98 - Message not compatible > > This cause indicates that the message received is not compatible with > the call state or the message type is non-existent or not implemented. > > In short it is a protocol error. Check out > http://www.telos-systems.com/?/techtalk/cause.htm for a complete lists > of causes and there meaning. > > > The SIP INVITE from the TNT to Asterisk contains no Caller Name > > information. > > > > It seems really odd to me that a Lucent TNT can not translate the caller > > ID Name info delivered by a Lucent 5ESS switch. > > > > On the same setup, if I connect another PRI device to it that emulates > > switch side signaling and includes the CNAM as a Display IE in the > > setup, the SIP invite is properly formatted and * receives the calling > > party name. > > > > Does anyone here have enough experience with ISDN PRI signaling to > > comment with some level of authority on this? > > Can you set a ISDN trace from your telco to your switch? I would be > curious to see what it looks like. > > Again, it looks like your telco's problem. Your best to ask them to > through a T
[Asterisk-Users] PrivacyManager not working Asterisk 1.0.9-BRIstuffed-0.2.0-RC8n
Hi list!
I'm trying to get PrivacyManager working but for some reason it always
thinks that CallerID is present (when it isn't). I get this on the
console:
== Primary D-Channel on span 1 up
-- Accepting voice call from '' to '0711234567' on channel 0/2, span 1
-- Executing Ringing("Zap/2-1", "") in new stack
-- Executing Zapateller("Zap/2-1", "answer|nocallerid") in new stack
-- Executing PrivacyManager("Zap/2-1", "") in new stack
-- CallerID Present: Skipping
-- Executing SetCIDNum("Zap/2-1", "") in new stack
-- Executing LookupCIDName("Zap/2-1", "") in new stack
I have this in my extensions.conf:
exten => 0711234567,1,Ringing
exten => 0711234567,2,Zapateller(answer|nocallerid)
exten => 0711234567,3,PrivacyManager
exten => 0711234567,4,SetCIDNum(${PRI_NETWORK_CID})
exten => 0711234567,5,LookupCIDName
exten => 0711234567,6,Dial(${EVERYONE},45,t)
exten => 0711234567,7,Answer
I need SetCIDNum(${PRI_NETWORK_CID}) to get CallerID working but I guess
it needn't be before Zapateller or PrivacyManager to trigger the
PrivacyManager.
Other question, would it be possible to use my own (custom recorded)
message instead of a standard prompt?
Thanks!!
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Re: [Asterisk-Users] Ring more than two isdn phones simultaneously
In your opinion, is there any chance that this situation will change in forseable future? Is anybody already working on such a "pseudo-channel" structure required for decoupling asterisk channels from the physical channels? If not, is there a significant interest in the asterisk community to do this? Arik Peter Svensson wrote: On Sun, 21 Aug 2005, Arik Funke wrote: If this is a limitation of asterisk, where is it located? In the chan_zap module? In zaphfc? I.e. would it help if I switched to mISDN? It is inherent in the channel-based structure of Asterisk. An audio channel is the basic measure used by applications such as Dial etc. This is shared by all channels as far as I know. One can imagine a special version of chan_zap that decouples the Asterisk channel entities from the actual B-channels. It would always generate a new fictitious asterisk channel structure and only link it to a real B-channel once the signaling indicated that a B-channel was required. I would be interested in how the commercial SS7 implementation for Asterisk works. SS7 would normally allow the audio paths to change in mid-call to potentially follow an altogether different route. Peter Peter Svensson <[EMAIL PROTECTED]> wrote: On Sat, 20 Aug 2005, Nico Giefing wrote: >> how many connection do you have from your asterisk to the old pbx? >> >> i think on 1 ISDN connection its only possible to let 2 phones ring, because >> 1 ISDN 2 channels... This is a limitation in Asterisk, not ISDN. Asterisk reserves a B-channel for each destination at the time of the CONNECT message. In the isdn world it is common to not actually allocate a B-channel until it is needed to carry audio. This also prevents Asterisk from letting the upstream switch select the B-channel on outgoing calls to the pstn. Asterisk is written this way since it uses the audio channel as the fundamental unit, with the D-channel as carrier of signalling for the individual B-channels. Another way to view ISDN is to consider the D-channel the fundamental unit, which can carry several audio streams as a side effect of the signalling. The first viewpoint resembles the traditional view of telephony as individual circuits, the second resembles the ISDN/SS7 view of the world. Changing Asterisk to be more ISDN-like is quite a lot of work. Peter >> >> - Original Message - >> From: "Arik Funke" <[EMAIL PROTECTED]> >> To: <[EMAIL PROTECTED]> >> Sent: Saturday, August 20, 2005 7:44 PM >> Subject: [Asterisk-Users] Ring more than two isdn phones simultaneously >> >> > >>> > I am using a HFC-S card in nt mode with zaphfc driver to connect an >>> > internal isdn bus. I would like to signal an incoming call on, let's >>> > say, 4 phones. Right now I use: >>> > >>> > Dial(Zap/g1/21&Zap/g1/22&Zap/g1/24&Zap/g1/23&Zap/g1/29,,t) >>> > >>> > where g1 are my two isdn channels provided by HFC-S card an the >>> > 21,22,etc my internal numbers. >>> > >>> > When the command is executed however, only the first two specified >>> > phones ring. Etc. with the first channel 21 ist called, with the second >>> > 22. How can I get asterisk to signal to all phones with just one isdn >>> > channel? I am trying to duplicate the setup I had with my old isdn pbx >>> > with did above trick just fine... Maybe somebody can help me configure >>> > asterisk appropriately? >>> > >>> > Cheers, >>> > Arik >>> > >>> > >>> > PS: I gave following a try but without success: >>> > Dial(Zap/g1/21-29,,t) >>> > Dial(Zap/g1/21+29,,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 >>> > > >> >> ___ >> 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 >> Peter ___ 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] On Network Usage from the CDR
Title: Message We want to use the accountcode in the CDR for billing and tracking total usage. We wanted to set the accountcode for calls coming into our network so we know which of our users to assign the usage to. But then when we receive an "on network" call we run into a problem with which accountcode to use for the call (our caller or our callee). It seems that it would be best to have two separate CDR entries for each "on network" call. I can imagine some ways of routing calls out and back into the box without using a valuable Zap channel. Or we could use the dst channel to track things but I am curious how have others dealt with this? Thanks Michael ___ 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] ZT_CHANCONFIG failed on channel 1 - WAS WORKING!!
Hi Everyone:
Problem Solved. Thanks to Matt, Paul and Rich for their excellent help!
It is always appreciated.
Here's the solution, for thsoe interested:
SuSE distributes zaptel drivers and the auto update referenced puts
drivers that may not be compatible with the current release on CVS (head
or stable) or whateve version you may have compiled and installed apart
from the disto.
Thus the incompatible symbols in loadable modules with no compile errors.
One set from the disto in the "extra" directory and the compiled set in
the "misc" directory under /lib/modules/{kernelname-version}/
It seems to me that this could conceivably cause Zaptel drivers to be
"flaky" as reported by Paul, when they aren't sufficiently out of sync
with symbol version error to fail in loading. For the complete story on
SuSE-Zaptel issues, please see:
http://www.voip-info.org/wiki-Asterisk+Linux+SuSE
Scott
On Sun, 21 Aug 2005 08:51:19 -0600, Scott Brown <[EMAIL PROTECTED]>
wrote:
Hi Paul:
Thanks very much for the suggestion. I don't understand why this just
began to happnen. I never
had problems before. Your suggestion has shed a lot of light on the
problem. Because wcfxs "disagrees" about the version of the symbols
listed below, I get the following unknown symbol problems for:
zt_receive
zt_qevent_lock
zt_ec_chunk
zt_transmit
zt_unregister
zt_hooksig
zt_register
With: "wcfxs: disagrees about version of symbol..."
BTW, I also did a full checkout (v1-0) and rebuild (deleting modules and
includes beforehand) of
astrisk zaptel and libpri. That didn't change anything.
Below is the resultant output from the -v modprobe and dmesg output. If
you or anyone else can help with this it would really be appreciated,
thanks:
# modprobe -v -n wctdm
insmod /lib/modules/2.6.11.4-21.8-default/kernel/lib/crc-ccitt.ko
insmod /lib/modules/2.6.11.4-21.8-default/misc/zaptel.ko
install /sbin/modprobe --ignore-install wcfxs && /sbin/ztcfg
# modprobe -v wctdm
insmod /lib/modules/2.6.11.4-21.8-default/kernel/lib/crc-ccitt.ko
insmod /lib/modules/2.6.11.4-21.8-default/misc/zaptel.ko
install /sbin/modprobe --ignore-install wcfxs && /sbin/ztcfg
insmod /lib/modules/2.6.11.4-21.8-default/extra/wcfxs.ko
FATAL: Error inserting wcfxs
(/lib/modules/2.6.11.4-21.8-default/extra/wcfxs.ko): Unknown symbol
in module, or unknown parameter (see dmesg)
FATAL: Error running install command for wcfxs
pertinent dmesg output:
zaptel: module not supported by Novell, setting U taint flag.
Zapata Telephony Interface Registered on major 196
wcfxs: module not supported by Novell, setting U taint flag.
Zapata Telephony Interface Unloaded
zaptel: module not supported by Novell, setting U taint flag.
Zapata Telephony Interface Registered on major 196
wcfxs: module not supported by Novell, setting U taint flag.
wcfxs: disagrees about version of symbol zt_receive
wcfxs: Unknown symbol zt_receive, st_info == 0x1
wcfxs: disagrees about version of symbol zt_qevent_lock
wcfxs: Unknown symbol zt_qevent_lock, st_info == 0x1
wcfxs: disagrees about version of symbol zt_ec_chunk
wcfxs: Unknown symbol zt_ec_chunk, st_info == 0x1
wcfxs: disagrees about version of symbol zt_transmit
wcfxs: Unknown symbol zt_transmit, st_info == 0x1
wcfxs: disagrees about version of symbol zt_unregister
wcfxs: Unknown symbol zt_unregister, st_info == 0x1
wcfxs: disagrees about version of symbol zt_hooksig
wcfxs: Unknown symbol zt_hooksig, st_info == 0x1
wcfxs: disagrees about version of symbol zt_register
wcfxs: Unknown symbol zt_register, st_info == 0x1
load_module: err 0xfffe (dont worry)
wcfxs: module not supported by Novell, setting U taint flag.
wcfxs: disagrees about version of symbol zt_receive
wcfxs: Unknown symbol zt_receive, st_info == 0x1
wcfxs: disagrees about version of symbol zt_qevent_lock
wcfxs: Unknown symbol zt_qevent_lock, st_info == 0x1
wcfxs: disagrees about version of symbol zt_ec_chunk
wcfxs: Unknown symbol zt_ec_chunk, st_info == 0x1
wcfxs: disagrees about version of symbol zt_transmit
wcfxs: Unknown symbol zt_transmit, st_info == 0x1
wcfxs: disagrees about version of symbol zt_unregister
wcfxs: Unknown symbol zt_unregister, st_info == 0x1
wcfxs: disagrees about version of symbol zt_hooksig
wcfxs: Unknown symbol zt_hooksig, st_info == 0x1
wcfxs: disagrees about version of symbol zt_register
wcfxs: Unknown symbol zt_register, st_info == 0x1
load_module: err 0xfffe (dont worry)
Thanks for the help.
Scott
On Sat, 20 Aug 2005 03:58:18 -0600, Paul Hewlett
<[EMAIL PROTECTED]> wrote:
On Saturday 20 August 2005 09:58, Scott Brown wrote:
Hi Matt:
That suggestion is possibly on the right track. It made me remember
that -
although I'm not using Fedora, but SuSE 9.3, that it went through an
automatic network update just recently. After that, I tried updating
the
Zaptel files from CVS and recompiling everything, but to no avail. The
same error still occured. I eliminated hardware by swapping out
Re: [Asterisk-Users] Ring more than two isdn phones simultaneously
On Sun, 21 Aug 2005, Arik Funke wrote: > If this is a limitation of asterisk, where is it located? In the > chan_zap module? In zaphfc? I.e. would it help if I switched to mISDN? It is inherent in the channel-based structure of Asterisk. An audio channel is the basic measure used by applications such as Dial etc. This is shared by all channels as far as I know. One can imagine a special version of chan_zap that decouples the Asterisk channel entities from the actual B-channels. It would always generate a new fictitious asterisk channel structure and only link it to a real B-channel once the signaling indicated that a B-channel was required. I would be interested in how the commercial SS7 implementation for Asterisk works. SS7 would normally allow the audio paths to change in mid-call to potentially follow an altogether different route. Peter > > Peter Svensson <[EMAIL PROTECTED]> wrote: > > On Sat, 20 Aug 2005, Nico Giefing wrote: > > >> how many connection do you have from your asterisk to the old pbx? > >> > >> i think on 1 ISDN connection its only possible to let 2 phones ring, > because > >> 1 ISDN 2 channels... > > > This is a limitation in Asterisk, not ISDN. Asterisk reserves a B-channel > for each destination at the time of the CONNECT message. In the isdn world > it is common to not actually allocate a B-channel until it is needed to > carry audio. This also prevents Asterisk from letting the upstream switch > select the B-channel on outgoing calls to the pstn. > > Asterisk is written this way since it uses the audio channel as the > fundamental unit, with the D-channel as carrier of signalling for the > individual B-channels. Another way to view ISDN is to consider the > D-channel the fundamental unit, which can carry several audio streams as a > side effect of the signalling. The first viewpoint resembles the > traditional view of telephony as individual circuits, the second resembles > the ISDN/SS7 view of the world. > > Changing Asterisk to be more ISDN-like is quite a lot of work. > > Peter > > > >> > >> - Original Message - > >> From: "Arik Funke" <[EMAIL PROTECTED]> > >> To: <[EMAIL PROTECTED]> > >> Sent: Saturday, August 20, 2005 7:44 PM > >> Subject: [Asterisk-Users] Ring more than two isdn phones simultaneously > >> > >> > > > >>> > I am using a HFC-S card in nt mode with zaphfc driver to connect an > >>> > internal isdn bus. I would like to signal an incoming call on, let's > >>> > say, 4 phones. Right now I use: > >>> > > >>> > Dial(Zap/g1/21&Zap/g1/22&Zap/g1/24&Zap/g1/23&Zap/g1/29,,t) > >>> > > >>> > where g1 are my two isdn channels provided by HFC-S card an the > >>> > 21,22,etc my internal numbers. > >>> > > >>> > When the command is executed however, only the first two specified > >>> > phones ring. Etc. with the first channel 21 ist called, with the > second > >>> > 22. How can I get asterisk to signal to all phones with just one isdn > >>> > channel? I am trying to duplicate the setup I had with my old > isdn pbx > >>> > with did above trick just fine... Maybe somebody can help me > configure > >>> > asterisk appropriately? > >>> > > >>> > Cheers, > >>> > Arik > >>> > > >>> > > >>> > PS: I gave following a try but without success: > >>> > Dial(Zap/g1/21-29,,t) > >>> > Dial(Zap/g1/21+29,,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 > >>> > > > > >> > >> ___ > >> 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 > >> > > > Peter ___ 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] Vtech/Vonage Wireless Phone system
Thanks, I just read all the literature on the Vtech website, and, I think you are exactly correct! Oh well, so much for cheap SIP wireless phones :-) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Rich Adamson Sent: Sunday, August 21, 2005 2:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Vtech/Vonage Wireless Phone system > I just stopped in at Best Buy here in CT, USA. I found an interesting > offering from Vtech there. It states it's a VOIP wireless phone system made > for the Vonage service. > > Here it is at their website: > http://www.vtechphones.com/vtechui/guide/dsp_voip_guide.cfm > > For $149, it comes with 2 (supposidly) VOIP 802.11b wireless phones, and a > 'broadband/Wireless' router. > > Does anyone know anything about these? Are they SIP phones? If so, I wonder > if it's possible to modify them for use with Asterisk? Do they need their > 'base' to talk, or is that nothing more than an 802.11b wireless router? > > I'm thinking of picking up a kit just to see if I can get them to work > nativly with Asterisk, but if anyone has any experience with them before I > do, I'd appreciate it.. I seen their ad in the paper this morning. It appeared the phones were the standard cordless (not 802.11b), and the base unit had pc wireless (802.11b) integrated with the linksys/sipura broadband stuff. You might want to check that out a little closer; but, fairly good return policy if the ads are a little misleading. ;) ___ 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] Broadvoice Issue
I did a quick google search of the lists site and couldn’t
find a definitive answer, so if it’s there, I apologize for asking again.
Starting about noon yesterday, I am no longer able to
send/receive calls via Broadvoice. When calling in, I get a fast busy, and when
calling out I get the following error:
-- Executing Dial("SIP/112-572a", "[EMAIL PROTECTED]")
in new stack
-- Called [EMAIL PROTECTED]
Aug 21 13:34:47 NOTICE[20742]: chan_sip.c:8648
handle_response: Failed to authenticate on INVITE to '"Mobile" ;tag=as124e3440'
== Spawn extension (agents, 78126631234, 1) exited
non-zero on 'SIP/112-572a'
I have the following in sip.conf:
register => [EMAIL PROTECTED]:password:[EMAIL PROTECTED]/
[sip.broadvoice.com]
type=peer
user=phone
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
fromuser=XX
secret=password
insecure=very
context=incoming
authname=XX
dtmfmode=inband
dtmf=inband
canreinvite=no
Does anyone know what I’m missing here? Everything was
working fine yesterday morning.
JT
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Re: [Asterisk-Users] "Not-Registered" Problem
Yes, and no. It would help to know what phones you're working with, what you've got in sip.conf, and what you've got in extensions.conf. -K > Hello, > What are some common reasons why a phone would report not registered > even when the extension has been setup through Asterisk(*) AND phone > username/password is correct? > > Joshua ___ 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] ViaTalk Down?
Is this a CVS-HEAD that was released AFTER 8/13? I ask because we're using that release and it's still deadlocking. Thanks, Sherwood McGowan ->-Original Message- ->From: [EMAIL PROTECTED] ->[mailto:[EMAIL PROTECTED] On Behalf Of ->Kevin P. Fleming ->Sent: Saturday, August 20, 2005 3:25 PM ->To: Asterisk Users Mailing List - Non-Commercial Discussion ->Subject: Re: [Asterisk-Users] ViaTalk Down? -> ->Sherwood McGowan wrote: -> ->> Anyone have a quickie answer as to why asterisk would suddenly just ->> stop responding? I was able to issue the restart command but I ->> couldn't do sip show peer and couldn't show channels, etc ->> This is very disconcerting -> ->Your SIP channel driver was deadlocked. This can happen for a ->number of reasons, but all of them are bad, and need to be fixed. -> ->Depending on the Asterisk version you are running, there may ->be some known situations under which they can occur; as best ->we can tell, there are none left in CVS HEAD related to chan_sip. ->___ ->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] Re: Asterisk 1.0.9 on SuSE 9.2 with ISDN BRI & zaphfc?
I would like to know how to install asterisk 1.0.9 with zaphfc working on a SuSE 9.2. I tried this: - The rpms with SuSE 9.2 are asterisk 1.0.6 - bristuff works, except for zaphfc, which doesn't compile. - The official asterisk download file doesn't contain isdn bri support Any ideas? Lars Dybdahl. ___ 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] Ring more than two isdn phones simultaneously
If this is a limitation of asterisk, where is it located? In the chan_zap module? In zaphfc? I.e. would it help if I switched to mISDN? Best regards, Arik Peter Svensson <[EMAIL PROTECTED]> wrote: On Sat, 20 Aug 2005, Nico Giefing wrote: >> how many connection do you have from your asterisk to the old pbx? >> >> i think on 1 ISDN connection its only possible to let 2 phones ring, because >> 1 ISDN 2 channels... This is a limitation in Asterisk, not ISDN. Asterisk reserves a B-channel for each destination at the time of the CONNECT message. In the isdn world it is common to not actually allocate a B-channel until it is needed to carry audio. This also prevents Asterisk from letting the upstream switch select the B-channel on outgoing calls to the pstn. Asterisk is written this way since it uses the audio channel as the fundamental unit, with the D-channel as carrier of signalling for the individual B-channels. Another way to view ISDN is to consider the D-channel the fundamental unit, which can carry several audio streams as a side effect of the signalling. The first viewpoint resembles the traditional view of telephony as individual circuits, the second resembles the ISDN/SS7 view of the world. Changing Asterisk to be more ISDN-like is quite a lot of work. Peter >> >> - Original Message - >> From: "Arik Funke" <[EMAIL PROTECTED]> >> To: <[EMAIL PROTECTED]> >> Sent: Saturday, August 20, 2005 7:44 PM >> Subject: [Asterisk-Users] Ring more than two isdn phones simultaneously >> >> > >>> > I am using a HFC-S card in nt mode with zaphfc driver to connect an >>> > internal isdn bus. I would like to signal an incoming call on, let's >>> > say, 4 phones. Right now I use: >>> > >>> > Dial(Zap/g1/21&Zap/g1/22&Zap/g1/24&Zap/g1/23&Zap/g1/29,,t) >>> > >>> > where g1 are my two isdn channels provided by HFC-S card an the >>> > 21,22,etc my internal numbers. >>> > >>> > When the command is executed however, only the first two specified >>> > phones ring. Etc. with the first channel 21 ist called, with the second >>> > 22. How can I get asterisk to signal to all phones with just one isdn >>> > channel? I am trying to duplicate the setup I had with my old isdn pbx >>> > with did above trick just fine... Maybe somebody can help me configure >>> > asterisk appropriately? >>> > >>> > Cheers, >>> > Arik >>> > >>> > >>> > PS: I gave following a try but without success: >>> > Dial(Zap/g1/21-29,,t) >>> > Dial(Zap/g1/21+29,,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 >>> > > >> >> ___ >> 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 >> Peter -- Peter Svensson ! Pgp key available by finger, fingerprint: <[EMAIL PROTECTED]> ! 8A E9 20 98 C1 FF 43 E3 07 FD B9 0A 80 72 70 AF Remember, Luke, your source will be with you... always... ___ 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] Vtech/Vonage Wireless Phone system
> I just stopped in at Best Buy here in CT, USA. I found an interesting > offering from Vtech there. It states it's a VOIP wireless phone system made > for the Vonage service. > > Here it is at their website: > http://www.vtechphones.com/vtechui/guide/dsp_voip_guide.cfm > > For $149, it comes with 2 (supposidly) VOIP 802.11b wireless phones, and a > 'broadband/Wireless' router. > > Does anyone know anything about these? Are they SIP phones? If so, I wonder > if it's possible to modify them for use with Asterisk? Do they need their > 'base' to talk, or is that nothing more than an 802.11b wireless router? > > I'm thinking of picking up a kit just to see if I can get them to work > nativly with Asterisk, but if anyone has any experience with them before I > do, I'd appreciate it.. I seen their ad in the paper this morning. It appeared the phones were the standard cordless (not 802.11b), and the base unit had pc wireless (802.11b) integrated with the linksys/sipura broadband stuff. You might want to check that out a little closer; but, fairly good return policy if the ads are a little misleading. ;) ___ 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] Problem with auto-attendant config, I think..
I never heard anything on the AMP list, so figured maybe someone here might
be able to help me sort this one out..
I was making some updates to my attendant config, which is really very
basic, and now incoming call processing stopped. Not sure exactly what the
heck happened, but figured maybe someone could help me with a clue as to what
broke. Now incoming calls are not being answered at all, no matter what I
do, which is quite annoying to say the least.
Here is some log output from the server:
Aug 20 11:16:46 VERBOSE[1617]: -- Starting simple switch on 'Zap/4-1'
Aug 20 11:16:47 DEBUG[1617]: Expression is '1'
Aug 20 11:16:47 VERBOSE[1617]: -- Executing GotoIf("Zap/4-1",
"1?from-pstn-reghours|s|1:") in new stack
Aug 20 11:16:47 VERBOSE[1617]: -- Goto (from-pstn-reghours,s,1)
Aug 20 11:16:47 DEBUG[1617]: Expression is '0'
Aug 20 11:16:47 VERBOSE[1617]: -- Executing GotoIf("Zap/4-1",
"0?from-pstn-reghours-nofax|s|1:2") in new stack
Aug 20 11:16:47 VERBOSE[1617]: -- Goto (from-pstn-reghours,s,2)
Aug 20 11:16:47 VERBOSE[1617]: -- Executing Answer("Zap/4-1", "") in new
stack
Aug 20 11:16:47 DEBUG[1617]: Took Zap/4-1 off hook
Aug 20 11:16:47 DEBUG[1617]: Enabled echo cancellation on channel 4
Aug 20 11:16:47 DEBUG[1617]: Engaged echo training on channel 4
Aug 20 11:16:47 VERBOSE[1617]: -- Executing Wait("Zap/4-1", "1") in new
stack
Aug 20 11:16:48 VERBOSE[1617]: -- Executing SetVar("Zap/4-1",
"intype=aa_2") in new stack
Aug 20 11:16:48 VERBOSE[1617]: -- Executing Cut("Zap/4-1",
"intype=intype|-|1") in new stack
Aug 20 11:16:48 DEBUG[1617]: Expression is '0'
Aug 20 11:16:48 VERBOSE[1617]: -- Executing GotoIf("Zap/4-1", "0?7:9") in
new stack
Aug 20 11:16:48 VERBOSE[1617]: -- Goto (from-pstn-reghours,s,9)
Aug 20 11:16:48 DEBUG[1617]: Expression is '0'
Aug 20 11:16:48 VERBOSE[1617]: -- Executing GotoIf("Zap/4-1", "0?10:12")
in new stack
Aug 20 11:16:48 VERBOSE[1617]: -- Goto (from-pstn-reghours,s,12)
Aug 20 11:16:48 DEBUG[1617]: Expression is '0'
Aug 20 11:16:48 VERBOSE[1617]: -- Executing GotoIf("Zap/4-1", "0?13:15")
in new stack
Aug 20 11:16:48 VERBOSE[1617]: -- Goto (from-pstn-reghours,s,15)
Aug 20 11:16:48 VERBOSE[1617]: -- Executing Goto("Zap/4-1", "aa_2|s|1") in
new stack
Aug 20 11:16:48 VERBOSE[1617]: -- Goto (aa_2,s,1)
Aug 20 11:16:48 WARNING[1617]: ast_yyerror(): syntax error: syntax error;
Input:
= ANSWER
^^
^
Aug 20 11:16:48 DEBUG[1617]: Expression is '0'
Aug 20 11:16:48 VERBOSE[1617]: -- Executing GotoIf("Zap/4-1", "0?4") in
new stack
Aug 20 11:16:48 DEBUG[1617]: Not taking any branch
Aug 20 11:16:48 VERBOSE[1617]: -- Executing Answer("Zap/4-1", "") in new
stack
Aug 20 11:16:48 VERBOSE[1617]: -- Executing Wait("Zap/4-1", "1") in new
stack
Aug 20 11:16:49 VERBOSE[1617]: -- Executing SetVar("Zap/4-1",
"DIR-CONTEXT=general") in new stack
Aug 20 11:16:49 VERBOSE[1617]: -- Executing DigitTimeout("Zap/4-1", "3")
in new stack
Aug 20 11:16:49 VERBOSE[1617]: -- Set Digit Timeout to 3
Aug 20 11:16:49 VERBOSE[1617]: -- Executing ResponseTimeout("Zap/4-1",
"7") in new stack
Aug 20 11:16:49 VERBOSE[1617]: -- Set Response Timeout to 7
Aug 20 11:16:49 VERBOSE[1617]: -- Executing BackGround("Zap/4-1",
"custom/aa_2") in new stack
Aug 20 11:16:49 DEBUG[1617]: Scheduling timer at 160 sample intervals
Aug 20 11:16:49 VERBOSE[1617]: -- Playing 'custom/aa_2' (language 'en')
Sure looks like it is trying to answer, but I keep hearing ringing on the
incoming side. If I look at my config for aa_2 I see:
[aa_2]
include => aa_2-custom
exten => fax,1,Goto(ext-fax,in_fax,1) ;
exten => h,1,Hangup() ;
exten => i,1,Playback(invalid) ;
exten => i,2,Goto(s,7) ;
include => ext-local
include => app-messagecenter
include => app-directory
exten => s,1,GotoIf($[${DIALSTATUS} = ANSWER]?4);
exten => s,2,Answer() ;
exten => s,3,Wait(1);
exten => s,4,SetVar(DIR-CONTEXT=general);
exten => s,5,DigitTimeout(3);
exten => s,6,ResponseTimeout(7) ;
exten => s,7,Background(custom/aa_2);
Heck and I thought I had this thing running well, and then it all breaks. If
anyone can point me in the right direction it would be most appreciated..
---
Howard Leadmon
http://www.leadmon.net
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Re: [Asterisk-Users] ZT_CHANCONFIG failed on channel 1 - WAS WORKING!!
> Hi Rich: > > Does the new distro tree issue explain the unresolved symbol references > noted in my last post? I don't know if SuSE's 9.3 network autoupdate > would have changed the tree structure. The unresolved symbols are (from > dmesg): > > zt_receive > zt_qevent_lock > zt_ec_chunk > zt_transmit > zt_unregister > zt_hooksig > zt_register > > With: "wcfxs: disagrees about version of symbol..." I don't use SuSE, so not 100% sure. Those sysbols are all related to the zaptel drivers, so I'd have to guess that something is messed up in that (as opposed to the kernel stuff). I don't recall if you mentioned using Stable or Head, but right now the cvs head appears to be a much better choice then Stable as a lot of effort has focused on getting Head cleaned up for a 1.2 release. I'm using cvs head with no problems right now; you might try that. ___ 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] Vtech/Vonage Wireless Phone system
Hi, I just stopped in at Best Buy here in CT, USA. I found an interesting offering from Vtech there. It states it's a VOIP wireless phone system made for the Vonage service. Here it is at their website: http://www.vtechphones.com/vtechui/guide/dsp_voip_guide.cfm For $149, it comes with 2 (supposidly) VOIP 802.11b wireless phones, and a 'broadband/Wireless' router. Does anyone know anything about these? Are they SIP phones? If so, I wonder if it's possible to modify them for use with Asterisk? Do they need their 'base' to talk, or is that nothing more than an 802.11b wireless router? I'm thinking of picking up a kit just to see if I can get them to work nativly with Asterisk, but if anyone has any experience with them before I do, I'd appreciate it.. - Andre ___ 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] Looking for Provider
www.teliax.com has treated me very well for about six months, and have lots of choices for DID (and 800) numbers. www.nufone.com handles calls very nicely and provides 800 numbers. But, their web site leaves a little to be desired, and they only use paypal. > -BEGIN PGP SIGNED MESSAGE- > Hash: SHA1 > > SIP and/or IAX would be nice. Residential. > > > Mark Phillips wrote: > > > I like Broadvoice but there are others. > > > > Do you want SIP or IAX termination? Business or residential? > > > > Mark > > > > Joshua Abbott wrote: > > > > > Hello > > I currently have internet service through MediaCom (Cable Internet) > > and need to find a VOIP provider that is compatible with Asterisk and > > Cable Internet. > > Any ideas? > > > > I'm in Missouri about 1.5 hours west of St Louis, MO in a town called > > Hermann (65041 zip code) > > > > > > Joshua > > >> > >> > ___ > 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 > > - -- > = > Joshua Abbott, Support Technician > http://www.successfulhosting.com/ > Direct Line: PENDING > Phone: (866) 494-5096 x1207 > > E-Fax: (419) 858-3241 > Alt E-Fax: (801) 217-1123 > [EMAIL PROTECTED] > = > The Success behind your web site! > = > -BEGIN PGP SIGNATURE- > Version: GnuPG v1.4.1 (GNU/Linux) > > iD8DBQFDCJYQeYokiwGEZc4RAtkNAJ4wjOWOyYigLfDr8NQWamBU013Z1gCeNXtq > sg/xpo114pDynqdVAh8IXpo= > =OeUp > -END PGP SIGNATURE- > > ---End of Original Message- ___ 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] TDM11B modprobe wcfxs fails
Rich Adamson wrote: > 2. I think the driver you want is wctdm (not wcfxs). I don't use the fxs modules, but the fxo modules use wctdm. Seems to me there was a change some time ago where the fxs modules are now supported from with wctdm. I'm not 100% sure though. In 1.0.x it's called wcfxs in post 1.0.x it is called wctdm. It's pretty easy to see what card goes with which driver. The info is in the README in the zaptel source directory. -- Eric Wieling * BTEL Consulting * 504-210-3699 x2120 ___ 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] TDM11B modprobe wcfxs fails
> I got my TDM11B and am trying to get it to work on my PC. > However, I'm having difficulty getting the wcfxs driver to > load. I've Googled this problem, and while there are others > who have ran into the same problem, none of the solutions > work for me. > > I would very much appreciate it if you could suggest things > I could try to resolve this problem. > > > Details follow. > > The PC is somewhat old - Tyan S2054, which has Intel i810 - > but I don't think this would be a problem. I > clean-installed Debian Sarge and am using 2.4.27 kernel. > > The TDM11B has its own IRQ (according to lspci -v). > > lspci -v shows the card as > > :01:04.0 Communication controller: Tiger Jet Network Inc. Tiger3XX > Modem/ISDN interface > > I'm following the instructions in > http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current_v1/docs-html/book1.html . > > I'm using the Asterisk/Zaptel software that's on the CD that > came with the TDM11B. It has version 1.0.7. > > The zaptel driver loads fine. The problem I'm having is > with the wcfxs driver. > > matsuri:~# modprobe zaptel > matsuri:~# modprobe wcfxs > /lib/modules/2.4.27/misc/wcfxs.o: init_module: No such device > Hint: insmod errors can be caused by incorrect module parameters, including > invalid IO or IRQ parameters. > You may find more information in syslog or the output from dmesg > /lib/modules/2.4.27/misc/wcfxs.o: insmod /lib/modules/2.4.27/misc/wcfxs.o > failed > /lib/modules/2.4.27/zaptel/wcfxs.o: init_module: No such device > Hint: insmod errors can be caused by incorrect module parameters, including > invalid IO or IRQ parameters. > You may find more information in syslog or the output from dmesg > /lib/modules/2.4.27/zaptel/wcfxs.o: insmod > /lib/modules/2.4.27/zaptel/wcfxs.o failed > /lib/modules/2.4.27/zaptel/wcfxs.o: insmod wcfxs failed > matsuri:~# > > I've tried all the PCI slots (4 of them). I get the same > error. > > The power cable is connected to the card. I've verified > that there is power to the cable and also tried different > power cable. > > The kernel used above is built by myself and it's not the > default kernel. (The default kernel that gets installed by > the Debian Installer couldn't even load the zaptel driver - > I get lots of missing references.) Two possible issues 1. The TDM card has several different revisions (rev e through h, I believe). If you have one of the later revisions, you may need zaptel software later then stable v1.07. 2. I think the driver you want is wctdm (not wcfxs). I don't use the fxs modules, but the fxo modules use wctdm. Seems to me there was a change some time ago where the fxs modules are now supported from with wctdm. I'm not 100% sure though. Also after doing a modprobe zaptel, do a lsmod to see what "is" loaded. Your likely to find all the appropriate drivers were loaded and you don't need to modprobe anything else. Look for something like this: wctdm 33728 4 wcfxo 13088 0 zaptel209028 19 wcusb,wctdm,wcfxo,wcte11xp,wct1xxp,wct4xxp,tor2 in the listing. ___ 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] Small office setup/using analog lines w/ Asterisk
I too have had trouble with FXO interfaces. I tried the Sipura SPA-3000 FXS/FXO device , X101p cards and a TDM11B card. None were satisfactory for my small office with 6 extensions and 3 lines. My longer term workaround was something that I setup just to bridge a perioud when I was taking down a test server and putting a poduction server online. I put provider based call forwarding on my POTS lines (only 2) and forwarded them to my IP based 800 number. This turned out so reliable that I have remained operating this way ever since, about 4 months now. People have remarked that the call quality is superb. I ever there is a problem with my DSL line or * server I simply turn off the call forwarding using a pair of analogue phones that I leave on the POTS lines for just such emergencies. This only addreses incomming calls. All outgoing calls are handled via IP through another termination provider. Michael On Sat, 20 Aug 2005 22:05:31 -0700, jennyw wrote: >Hi, > >We recently tried installing Asterisk for a small office. We figured the >safest way to go would be to buy from someone who sold equipment >specifically for Asterisk and to use a consultant that they >recommended. However ... it didn't turn out so great. Sound quality is >terrible -- the echo is pretty bad, and there are popping noises, too. >Callers say that people on the Asterisk end sound very faint, while <,snip> >One of the next tests will be using Asterisk with a VoIP provider to see >what the sound quality is like with digital on both ends. PRI sounds >like it'd be even better, but for an office w/ 5 people, it sounds >pretty expensive. How do other people do this? > >Thanks in advance for any pointers! > >Jen > > >___ >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 > > -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 fwd 54245 ___ 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] Asterisk(*) on a Cobalt RaQ2?
Sure, this can work. I have an Asterisk install in my home office running about the same hardware. Our is a newer mini-itx system booting Astlinux from a compact flash card. Just don't bother with any form of transcoding, especially G.729a...you don't have the cpu power for more than 2 channels of transcode with out other processes suffering badly. Stay G.711 all the way and you can have multiple calls ongoing, VM, even conferencing without any trouble. The wiki has good info on running * on small systems. Michael On Sun, 21 Aug 2005 01:25:32 -0500, Joshua Abbott wrote: >-BEGIN PGP SIGNED MESSAGE- >Hash: SHA1 > >Can Asterisk(*) be installed on this and if so are the setup >instructions any different? >I have a client that's asking. > >Specs: 533Mhz, 512MB RAM, 10GB HDD, Cobalt 4 OS > >- -- >= >Joshua Abbott, Support Technician >http://www.successfulhosting.com/ >Direct Line: PENDING >Phone: (866) 494-5096 x1207 > >E-Fax: (419) 858-3241 >Alt E-Fax: (801) 217-1123 >[EMAIL PROTECTED] >= >The Success behind your web site! >= >-BEGIN PGP SIGNATURE- >Version: GnuPG v1.4.1 (GNU/Linux) > >iD8DBQFDCB5ceYokiwGEZc4RAj88AKCK0uqXTgqm2Yo1eUJAW+CGrvrUqwCgxf7e >hpMjXYmHGDb0x2lPXrOE19w= >=rKsn >-END PGP SIGNATURE- > > -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 fwd 54245 ___ 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 voice one way only
The easiest thing is probably to get a card that is more widely supported. Any cheap pci HFC-S card will do, they are sold for anthything between 9 and 15 eur. With an hfc-s card you can then use bristuff or chan_capi On Sun, 21 Aug 2005, Klemens Kasemaa wrote: hi PSTN <--> [Teles ISDN / Asterisk] <--> SIP client When call is made through ISDN, no matter if taken from PSTN or Asterisk side, person in PSTN side can hear perfectly but in Asterisk side I only hear a very scrambled or very low quality voice, words repeated several times. Same is with echo test (call taken from PSTN) Get a CAPI module for your Teles and try chan_capi-cm from http://sourceforge.net/projects/chan-capi/ Accordingly capi.org this card does have capi support but not under linux. Because of it's a ISA card, I can't use zaphfc also. So any help is appreciated. with rgrds, klem ___ 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
Re: [Asterisk-Users] ZT_CHANCONFIG failed on channel 1 - WAS WORKING!!
Hi Rich: Does the new distro tree issue explain the unresolved symbol references noted in my last post? I don't know if SuSE's 9.3 network autoupdate would have changed the tree structure. The unresolved symbols are (from dmesg): zt_receive zt_qevent_lock zt_ec_chunk zt_transmit zt_unregister zt_hooksig zt_register With: "wcfxs: disagrees about version of symbol..." Scott On Sat, 20 Aug 2005 06:11:53 -0600, Rich Adamson <[EMAIL PROTECTED]> wrote: Seems the latest distro's have changed the layout of the linux source tree needed to compile the zaptel stuff. I'm using FC3, upgraded the kernel, and have the same issue. Was able to install new sources, but they too are completely different tree layout compared to earlier stuff. The same is apparently happening with other distro's as well. There has been a bug item open for last several weeks relative to reworking the make files for these items. That suggestion is possibly on the right track. It made me remember that - although I'm not using Fedora, but SuSE 9.3, that it went through an automatic network update just recently. After that, I tried updating the Zaptel files from CVS and recompiling everything, but to no avail. The same error still occured. I eliminated hardware by swapping out a working TDM400 with the same FXS/FXO configuration. The same error occurs. The SuSE update may have moved some of the required files, although there are no complaints during the build and I can't determine what may have moved. I have still present and installed Bison 1.875, OpenSSL and zlib-devel, and of course Linux source for this SuSE disto. I'm completely faklempt! Can someone shed light on this delima?? Thanks so much if you can. I want my As-terisk back!!! It was working, damnit. Thanks, Matt for your suggestion. Scott At 09:30 PM 8/19/2005, you wrote: >[EMAIL PROTECTED] wrote: > > > > Hi: > > I hope that someone can help with this problem that came up suddenly. I > >Did you upgrade Fedora Core? > >Check if the udev files still contain the required entries (normally fedora >copies the old ones to 50-udev-rules.old and makes new ones). ___ 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 -- Using Opera's revolutionary e-mail client: http://www.opera.com/mail/ ___ 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] ZT_CHANCONFIG failed on channel 1 - WAS WORKING!!
Hi Paul: Thanks very much for the suggestion. I don't understand why this just began to happnen. I never had problems before. Your suggestion has shed a lot of light on the problem. Because wcfxs "disagrees" about the version of the symbols listed below, I get the following unknown symbol problems for: zt_receive zt_qevent_lock zt_ec_chunk zt_transmit zt_unregister zt_hooksig zt_register With: "wcfxs: disagrees about version of symbol..." BTW, I also did a full checkout (v1-0) and rebuild (deleting modules and includes beforehand) of astrisk zaptel and libpri. That didn't change anything. Below is the resultant output from the -v modprobe and dmesg output. If you or anyone else can help with this it would really be appreciated, thanks: # modprobe -v -n wctdm insmod /lib/modules/2.6.11.4-21.8-default/kernel/lib/crc-ccitt.ko insmod /lib/modules/2.6.11.4-21.8-default/misc/zaptel.ko install /sbin/modprobe --ignore-install wcfxs && /sbin/ztcfg # modprobe -v wctdm insmod /lib/modules/2.6.11.4-21.8-default/kernel/lib/crc-ccitt.ko insmod /lib/modules/2.6.11.4-21.8-default/misc/zaptel.ko install /sbin/modprobe --ignore-install wcfxs && /sbin/ztcfg insmod /lib/modules/2.6.11.4-21.8-default/extra/wcfxs.ko FATAL: Error inserting wcfxs (/lib/modules/2.6.11.4-21.8-default/extra/wcfxs.ko): Unknown symbol in module, or unknown parameter (see dmesg) FATAL: Error running install command for wcfxs pertinent dmesg output: zaptel: module not supported by Novell, setting U taint flag. Zapata Telephony Interface Registered on major 196 wcfxs: module not supported by Novell, setting U taint flag. Zapata Telephony Interface Unloaded zaptel: module not supported by Novell, setting U taint flag. Zapata Telephony Interface Registered on major 196 wcfxs: module not supported by Novell, setting U taint flag. wcfxs: disagrees about version of symbol zt_receive wcfxs: Unknown symbol zt_receive, st_info == 0x1 wcfxs: disagrees about version of symbol zt_qevent_lock wcfxs: Unknown symbol zt_qevent_lock, st_info == 0x1 wcfxs: disagrees about version of symbol zt_ec_chunk wcfxs: Unknown symbol zt_ec_chunk, st_info == 0x1 wcfxs: disagrees about version of symbol zt_transmit wcfxs: Unknown symbol zt_transmit, st_info == 0x1 wcfxs: disagrees about version of symbol zt_unregister wcfxs: Unknown symbol zt_unregister, st_info == 0x1 wcfxs: disagrees about version of symbol zt_hooksig wcfxs: Unknown symbol zt_hooksig, st_info == 0x1 wcfxs: disagrees about version of symbol zt_register wcfxs: Unknown symbol zt_register, st_info == 0x1 load_module: err 0xfffe (dont worry) wcfxs: module not supported by Novell, setting U taint flag. wcfxs: disagrees about version of symbol zt_receive wcfxs: Unknown symbol zt_receive, st_info == 0x1 wcfxs: disagrees about version of symbol zt_qevent_lock wcfxs: Unknown symbol zt_qevent_lock, st_info == 0x1 wcfxs: disagrees about version of symbol zt_ec_chunk wcfxs: Unknown symbol zt_ec_chunk, st_info == 0x1 wcfxs: disagrees about version of symbol zt_transmit wcfxs: Unknown symbol zt_transmit, st_info == 0x1 wcfxs: disagrees about version of symbol zt_unregister wcfxs: Unknown symbol zt_unregister, st_info == 0x1 wcfxs: disagrees about version of symbol zt_hooksig wcfxs: Unknown symbol zt_hooksig, st_info == 0x1 wcfxs: disagrees about version of symbol zt_register wcfxs: Unknown symbol zt_register, st_info == 0x1 load_module: err 0xfffe (dont worry) Thanks for the help. Scott On Sat, 20 Aug 2005 03:58:18 -0600, Paul Hewlett <[EMAIL PROTECTED]> wrote: On Saturday 20 August 2005 09:58, Scott Brown wrote: Hi Matt: That suggestion is possibly on the right track. It made me remember that - although I'm not using Fedora, but SuSE 9.3, that it went through an automatic network update just recently. After that, I tried updating the Zaptel files from CVS and recompiling everything, but to no avail. The same error still occured. I eliminated hardware by swapping out a working TDM400 with the same FXS/FXO configuration. The same error occurs. The SuSE update may have moved some of the required files, although there are no complaints during the build and I can't determine what may have moved. Currently I am doing the following on SuSE : First reboot the PC with asterisk disabled. This will force the creation of the devices during boot from the /etc/udev/rules.d files. Try modprobing : modprobe -v -n wctdm This does nothing but tells you what would happen. If your modprobe.d/zaptel file is correct the the output from this command will be loading of zaptel,wcfxs and an execution of ztcfg. In other words you do not have to modprobe more than one module - dependencies are sorted by the modprobe.d/zaptel file. If you want -vv on the ztcfg file edit modprobe.d/zaptel. I remember from the wiki somewhere that one must not execute ztcfg more than once and this will happen if you modprobe zaptel and then wctdm and then e
[Asterisk-Users] "Not-Registered" Problem
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hello, What are some common reasons why a phone would report not registered even when the extension has been setup through Asterisk(*) AND phone username/password is correct? Joshua - -- = Joshua Abbott, Support Technician http://www.successfulhosting.com/ Direct Line: PENDING Phone: (866) 494-5096 x1207 E-Fax: (419) 858-3241 Alt E-Fax: (801) 217-1123 [EMAIL PROTECTED] = The Success behind your web site! = -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.1 (GNU/Linux) iD8DBQFDCJ5ieYokiwGEZc4RAlsLAJ9TuT9Xr5OqhrUtvDp3zwtu7Lt5fQCggvno uCTezwNe8PXSYftSQ1dzUQc= =co8L -END PGP SIGNATURE- begin:vcard fn:Joshua Abbott (Successful Hosting) n:Abbott;Joshua org:Successful Hosting;Support adr:3009 Avenue J;;Attn: Joshua Abbott;Brooklyn;NY;11210;USA email;internet:[EMAIL PROTECTED] title:Technical Support Representative tel;work:+1 (866) 494-5096 x1207 tel;fax:+1 (419) 858-3241 note:Alt E-Fax: (801) 217-1123 x-mozilla-html:FALSE url:http://www.successfulhosting.com/ version:2.1 end:vcard ___ 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] Small office setup/using analog lines w/ Asterisk
On Sunday 21 August 2005 01:05, jennyw wrote: > up too high). The reseller and the consultant both say that the most > likely cause for this is using Digium cards w/ analog phone lines. > Apparently, they say, sound quality can be pretty bad. The reseller/consultant aren't worth the money you paid them, then. Interfacing to plain-old analog lines can be problematic. The TDM400P FXO modules are tunable but it takes time and testing to get it right, not playing about willy-nilly with settings in an attempt to solve the problem through entropic little adjustments without a clear idea of what they do and how they work. Asterisk is a very difficult application on a system. Interfacing to anything outside the computer, whether it be an analog telephone line, a local SIP phone or a remote VOIP provider requires that the system's ability to access its resources reliably and with repeatable access times. As simple as this sounds it is a very difficult problem and the #1 reason why VOIP is so difficult to roll out on commodity hardware. You simply can't use any old system and any old network card and any old network gear (router/firewall, switches, etc.) and get good results. > nothing has worked well. So I wanted to ask others ... has anyone had > good luck with using analog phone lines and Asterisk? Especially with > Digium cards (we use the TDM400P)? Although from reading articles on the > net it sounds like people do have a lot of echo problems, it also sounds > like some people are using analog phone lines with some success. The echo problems are almost always due to one of two things: poor line tuning or crappy base hardware (computer). Now the older version of the TDM cards and FXO modules specifically had issues, but they have, to my knowledge, all been resolved. I used to recommend a T1 card + channel bank (Adit600) even for a couple channels, but nowadays I have no compunctions in recommending the TDM400 and FXO modules. > echocancelwhenbridged, txgain, and rxgain in zapata.conf. I've heard > from the reseller that what might work better is to trade the Digium > cards in for VegaStream gateway. It's more expensive, but apparently has > a DSP built in that should increase voice quality. Of course, they say > there are no guarantees with this. They also mentioned (after the fact) > that Asterisk systems don't necessarily save money. So far, the Find a new reseller, and post their name here so we can all avoid them. I've rolled out numerous asterisk installations with good success. As I mentioned earlier, the trick is measured, controlled tests and methodical experimentation. http://lists.digium.com/pipermail/asterisk-users/2004-November/071301.html is the method I use to tune the line, but there is more to it now since there is the fxotune application which is used to set up the FIR filter on the FXO card's DAA. If you're using a SIP phone to test with, put a second network card in the Asterisk box and plug the phone directly into it (with a crossover cable) and adjust the Asterisk settings to see and use the second card. This will determine if your network is causing issues by eliminating it from the equation. Make sure you use a decent network card (I love the Intel eepro/100 cards myself). Also, if you're going out through a VOIP provider, make damn sure you either use a dedicated DSL link (my personal recommendation) or make really sure that your router can properly tag and prioritize outgoing traffic, and that it is also doing its best to prevent the "other side" of the link from flooding your incoming pipe. http://www.mixdown.ca/~andrew/dump/rc.tc is the script I use with good success. As far as Asterisk being more expensive than other systems... doubtful. You can get a cheap Nortel 3x8 for cheap, sure, but then its limitations will have you buying a small MICS... Now add their $4000 voicemail system, $500 trunk cards for four FXO channels... oh wait, you want caller-id on those? $600 then... oh wait, you want VOIP on it? $2500 here, $500 there, $1000 the other place... Make sure you're comparing apples to apples. I feel that Asterisk runs *very* well on most hardware I've thrown it at, and it is far far far more configurable than any proprietary KSU or PBX, and a damn sight cheaper than *ANY* PBX out there. > One of the next tests will be using Asterisk with a VoIP provider to see > what the sound quality is like with digital on both ends. PRI sounds > like it'd be even better, but for an office w/ 5 people, it sounds > pretty expensive. How do other people do this? Yes, an office with 5 people (probably only two POTS lines I am guessing) is not really a good choice for PRI. ISDN BRI if you can get it would be better, or simply finding a trusted VOIP provider and getting a DID from them would be easiest, but I would then recommend two DSL providers on the same POTS line (it's all PPPoE anyway) for failover. -A. _
RE: [Asterisk-Users] Re: [Asterisk-Dev] IM patch
What about asterisk chan_sip and IM +presence !!! Harry --- "Olle E. Johansson" <[EMAIL PROTECTED]> a écrit : > harry gaillac wrote: > > Hello, > > > > I patched asterisk cvs head sources with > > > http://juraj.bednar.sk/work/software/asterisk/messaging/ > > and presnce patch without success. > > > > asterisk send "405 method not allowed" to sender. > > I use polycom ip300. > THat is a response to the polycom's PUBLISH request, > a method that > is not implemented. > > /O > ___ > Asterisk-Dev mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-dev > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-dev > > ___ > 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 > ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ 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] Small office setup/using analog lines w/
On the clicking front, it could still be packet loss. I recently (just this week in fact) solved a clicking problem my client was having. turned out to be interference from his wireless network bridge. Both it and his Cisco ATA186 were sitting next to each other. I put about 3 feet of space between them and solved the problem. I think what was happening was that the radio was interfering with the data on the cable causing loss/bad checksums/whatever. That's just my guess. What would I know? I'm only an Extra Class ham radio operator. Mark [EMAIL PROTECTED] wrote: Jenny - I'm glad I'm not the only one! I just installed Asterisk on Friday and I spent all day trying to de-gremlin my system. I'm glad I'm doing it for myself and we haven't switched from our legacy system yet, but I have a potential client that wants to see how well I can implement this system for myself... (Speaking of which, don't forget that your supplier and consultant probably make more off of other phone systems and a lot of old-line suppliers are kind of afraid of Asterisk. There were some big companies that lost out when the State of Oregon's court system went to Asterisk for their teleconferencing needs... 600 MeetMe sessions a day does say something, I think.) First, it would help -me- to know precisely what hardware you're using (including network... switches and cards) and what phones you're using. You can email me off-list if you want. I'm personally using a AMD Sempron-based system with a TDM-04B (4x FXO) in it, routing out through an IntelPro 10/100 card. The data switch is currently a crappy D-Link, but more on that and the network card in a second. The other end is one soft phone on Linux and two SPA-841 hardware phones. When I first installed, the echo was terrible. Tweaking around with the echo settings in zapata.conf endlessly didn't do a whole lot. So, using my go-go-gadget troubleshooting instincts (and my handy-dandy go-go-gadget credit card) I figured the problem might be network latency ... so I replaced the crappy realtek network card with a high-quality Intel one, and borrwed a Dell managed switch from a client for the day. Voila, a few more tweaks to the echo learning rate and my echo problems went away, and I spent 45 minutes learning about my sister's love life in the wilds of downtown San Francisco. (Note to self: Don't use family to test telecom systems.) The clicking persisted, though. I thought it was the phone at first, but if you're experiencing it too... well, apparently it's not. I searched the archives and couldn't find any other reference to clicking noises, so I'm at a loss but I'm hoping it's wiring-based and heading back to the office tomorrow to try and solve it. I think you're right that Digium is focusing on their T-1/E-1 market. I've noticed it when I called with a question, and that really ticks me off. It might be a decent, sound business decision based on profits for the different cards, but it's leaving a lot of us smaller business operators -- those who might take great experiences with Digium and Asterisk products into potential large customers -- swinging in the wind. -Karl Katzke Streetlamp Software Solutions http://www.streetlampsoftware.com Hi, We recently tried installing Asterisk for a small office. We figured the safest way to go would be to buy from someone who sold equipment specifically for Asterisk and to use a consultant that they recommended. However ... it didn't turn out so great. Sound quality is terrible -- the echo is pretty bad, and there are popping noises, too. Callers say that people on the Asterisk end sound very faint, while people on the Asterisk end hear people maybe too loundly (might be related to the popping noises -- sounds like when you have stereo turned up too high). The reseller and the consultant both say that the most likely cause for this is using Digium cards w/ analog phone lines. Apparently, they say, sound quality can be pretty bad. I called Digium and they gave me some suggestions for settings, but nothing has worked well. So I wanted to ask others ... has anyone had good luck with using analog phone lines and Asterisk? Especially with Digium cards (we use the TDM400P)? Although from reading articles on the net it sounds like people do have a lot of echo problems, it also sounds like some people are using analog phone lines with some success. FYI, what I've mainly done is try changing echotraining, echocancel, echocancelwhenbridged, txgain, and rxgain in zapata.conf. I've heard from the reseller that what might work better is to trade the Digium cards in for VegaStream gateway. It's more expensive, but apparently has a DSP built in that should increase voice quality. Of course, they say there are no guarantees with this. They also mentioned (after the fact) that Asterisk systems don't necessarily save money. So far, the experience has been very frustrating and I'd love to hear some success stories from others (or more in
Re: [Asterisk-Users] Looking for Provider
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 SIP and/or IAX would be nice. Residential. Mark Phillips wrote: > I like Broadvoice but there are others. > > Do you want SIP or IAX termination? Business or residential? > > Mark > > Joshua Abbott wrote: > > Hello > I currently have internet service through MediaCom (Cable Internet) > and need to find a VOIP provider that is compatible with Asterisk and > Cable Internet. > Any ideas? > > I'm in Missouri about 1.5 hours west of St Louis, MO in a town called > Hermann (65041 zip code) > > > Joshua >> >> ___ 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 - -- = Joshua Abbott, Support Technician http://www.successfulhosting.com/ Direct Line: PENDING Phone: (866) 494-5096 x1207 E-Fax: (419) 858-3241 Alt E-Fax: (801) 217-1123 [EMAIL PROTECTED] = The Success behind your web site! = -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.1 (GNU/Linux) iD8DBQFDCJYQeYokiwGEZc4RAtkNAJ4wjOWOyYigLfDr8NQWamBU013Z1gCeNXtq sg/xpo114pDynqdVAh8IXpo= =OeUp -END PGP SIGNATURE- begin:vcard fn:Joshua Abbott (Successful Hosting) n:Abbott;Joshua org:Successful Hosting;Support adr:3009 Avenue J;;Attn: Joshua Abbott;Brooklyn;NY;11210;USA email;internet:[EMAIL PROTECTED] title:Technical Support Representative tel;work:+1 (866) 494-5096 x1207 tel;fax:+1 (419) 858-3241 note:Alt E-Fax: (801) 217-1123 x-mozilla-html:FALSE url:http://www.successfulhosting.com/ version:2.1 end:vcard ___ 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] Looking for Provider
I like Broadvoice but there are others. Do you want SIP or IAX termination? Business or residential? Mark Joshua Abbott wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hello I currently have internet service through MediaCom (Cable Internet) and need to find a VOIP provider that is compatible with Asterisk and Cable Internet. Any ideas? I'm in Missouri about 1.5 hours west of St Louis, MO in a town called Hermann (65041 zip code) Joshua -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.1 (GNU/Linux) iD8DBQFDB+6deYokiwGEZc4RAmjiAKCXEqn/X4b8blsXsqF7YM4//9COqACgp2Of IUOztI66i882+yCncoSYALk= =L8Pk -END PGP SIGNATURE- ___ 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 -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com ___ 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 voice one way only
hi > > PSTN <--> [Teles ISDN / Asterisk] <--> SIP client > > > > When call is made through ISDN, no matter if taken from PSTN or > > Asterisk side, person in PSTN side can hear perfectly but in Asterisk > > side I only hear a very scrambled or very low quality voice, words > > repeated several times. Same is with echo test (call taken from PSTN) > > Get a CAPI module for your Teles and try chan_capi-cm from > http://sourceforge.net/projects/chan-capi/ Accordingly capi.org this card does have capi support but not under linux. Because of it's a ISA card, I can't use zaphfc also. So any help is appreciated. with rgrds, klem ___ 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] Warning Unable to allocate socket
Kamran Ahmad wrote: i m getting follwing messages in asterisk-1.0.9 what is the reason calls are not going out. can u pls tel me how to solve this Aug 20 13:06:09 WARNING[7706]: rtp.c:829 ast_rtcp_new: Unable to allocate socket: Too many open files Aug 20 13:06:09 WARNING[7706]: channel.c:311 ast_channel_alloc: Alert pipe creation failed! Aug 20 13:06:09 WARNING[7706]: chan_sip.c:2081 sip_new: Unable to allocate channel structure Aug 20 13:06:09 NOTICE[7706]: chan_sip.c:7469 handle_request: Unable to create/find channel Aug 20 13:06:22 WARNING[7706]: acl.c:216 ast_lookup_iface: Unable to get IP of eth0: Bad file descriptor stop asterisk, then start asterisk. Asterisk (at least 1.0.9) can leak file descriptors in some situations and you should restart Asterisk every once in a while. ___ 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: Asterisk-Users Digest, Vol 13, Issue 131
On Sunday 21 August 2005 02:24, Nguyen Trung Tin wrote: > Hello > > how to calculator billing exactly when IAX accept the call, my configure > > --> telco ---> asterisk --> ACD --> IAX The phone company does not bill for talk time but for use time. You used the phone network for 33 seconds. Besides if you make that call you will be billed according to their billing system. Which may bill in a number of ways. X amount for the first minute, then Y for every N seconds. That can typically translate to 60 second, 6 second or 1 second billing increments. -- List Manager Network Voice Communications, Inc. netwvcom.com ___ 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] Asterisk 1.0.9 on SuSE 9.2 with ISDN BRI & zaphfc?
I would like to know how to install asterisk 1.0.9 with zaphfc working on a SuSE 9.2. I tried this: - The rpms with SuSE 9.2 are asterisk 1.0.6 - bristuff works, except for zaphfc, which doesn't compile. - The official asterisk download file doesn't contain isdn bri support Any ideas? Lars Dybdahl. ___ 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] Small office setup/using analog lines w/ Asterisk
People, please cut down the original post in your replies. It's wasting space, bandwidth and time. -- List Manager ___ 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] Searching For a Voip Provider
On Thursday 18 August 2005 14:55, chawki hammoud wrote: > Hi: > > Please advice me of a voip provider with reasonable > reseller program. I was using voipjet and it has a lot > of problems. > > Did anyone experienced asteriskout.com service? They > have good prices. What you may want to do is to subscribe to asterisk-biz where you can find several carriers. -- List Manager Network Voice Communications, Inc. netwvcom.com ___ 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] Asterisk(*) on a Cobalt RaQ2?
On Sunday 21 August 2005 02:25, Joshua Abbott wrote: > Can Asterisk(*) be installed on this and if so are the setup > instructions any different? > I have a client that's asking. > > Specs: 533Mhz, 512MB RAM, 10GB HDD, Cobalt 4 OS As long as you don't expect to run more than a couple of calls at a time through it, it will work. You don't have much CPU power to go around. I use a similar box at my home and it's OK for that, but hardly for a business. Pickup this script and you'll have an easy install. wget szmidt.org/asterisk/asterisk-update.sh Set the execute bit and run it. To get a specific version you'll modify the version variable at the top. -- List Manager Network Voice Communications, Inc. netwvcom.com ___ 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] Small office setup/using analog lines w/ Asterisk
Jennyw, I have setup about 8 Asterisk systems with The TDM400p boards in them. Yes allot of them had at the beginning some echo and other things. But I have been able to work and get them fixed. 1) Make sure your motherboard is able to assign it's own IRQ for the board. This is one of the most important things. 2) There is a tool on the system that lets you set the txgain and rxgain. It's called ztmonitor which you can use to see how the volume is setup then you can make some adjustments. 3) It's important to make sure your system has good network card. I have had some problems with Realtech with echo. 4) You also need to make sure you have the phones on connect to a switch. 5) Asterisk system does save allot of money in the long run. I am sorry that the person you got was not able to help. But I know that the TDM400p boards have there well critic's but they do work. 6) For us to give you more help we are going to need to know more about you system. What is the server your using? What phones? How is your network setup? If you want you can email me directly. I will try to help you out with your setup. Ariel Batista jennyw wrote: Hi, We recently tried installing Asterisk for a small office. We figured the safest way to go would be to buy from someone who sold equipment specifically for Asterisk and to use a consultant that they recommended. However ... it didn't turn out so great. Sound quality is terrible -- the echo is pretty bad, and there are popping noises, too. Callers say that people on the Asterisk end sound very faint, while people on the Asterisk end hear people maybe too loundly (might be related to the popping noises -- sounds like when you have stereo turned up too high). The reseller and the consultant both say that the most likely cause for this is using Digium cards w/ analog phone lines. Apparently, they say, sound quality can be pretty bad. I called Digium and they gave me some suggestions for settings, but nothing has worked well. So I wanted to ask others ... has anyone had good luck with using analog phone lines and Asterisk? Especially with Digium cards (we use the TDM400P)? Although from reading articles on the net it sounds like people do have a lot of echo problems, it also sounds like some people are using analog phone lines with some success. FYI, what I've mainly done is try changing echotraining, echocancel, echocancelwhenbridged, txgain, and rxgain in zapata.conf. I've heard from the reseller that what might work better is to trade the Digium cards in for VegaStream gateway. It's more expensive, but apparently has a DSP built in that should increase voice quality. Of course, they say there are no guarantees with this. They also mentioned (after the fact) that Asterisk systems don't necessarily save money. So far, the experience has been very frustrating and I'd love to hear some success stories from others (or more info on what I can realistically expect from an Asterisk system)! And, of course, some ideas on how I can get things to work better. One of the next tests will be using Asterisk with a VoIP provider to see what the sound quality is like with digital on both ends. PRI sounds like it'd be even better, but for an office w/ 5 people, it sounds pretty expensive. How do other people do this? Thanks in advance for any pointers! Jen ___ 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] TDM11B modprobe wcfxs fails
Hi, I got my TDM11B and am trying to get it to work on my PC. However, I'm having difficulty getting the wcfxs driver to load. I've Googled this problem, and while there are others who have ran into the same problem, none of the solutions work for me. I would very much appreciate it if you could suggest things I could try to resolve this problem. Details follow. The PC is somewhat old - Tyan S2054, which has Intel i810 - but I don't think this would be a problem. I clean-installed Debian Sarge and am using 2.4.27 kernel. The TDM11B has its own IRQ (according to lspci -v). lspci -v shows the card as :01:04.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface I'm following the instructions in http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current_v1/docs-html/book1.html. I'm using the Asterisk/Zaptel software that's on the CD that came with the TDM11B. It has version 1.0.7. The zaptel driver loads fine. The problem I'm having is with the wcfxs driver. matsuri:~# modprobe zaptel matsuri:~# modprobe wcfxs /lib/modules/2.4.27/misc/wcfxs.o: init_module: No such device Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg /lib/modules/2.4.27/misc/wcfxs.o: insmod /lib/modules/2.4.27/misc/wcfxs.o failed /lib/modules/2.4.27/zaptel/wcfxs.o: init_module: No such device Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg /lib/modules/2.4.27/zaptel/wcfxs.o: insmod /lib/modules/2.4.27/zaptel/wcfxs.o failed /lib/modules/2.4.27/zaptel/wcfxs.o: insmod wcfxs failed matsuri:~# I've tried all the PCI slots (4 of them). I get the same error. The power cable is connected to the card. I've verified that there is power to the cable and also tried different power cable. The kernel used above is built by myself and it's not the default kernel. (The default kernel that gets installed by the Debian Installer couldn't even load the zaptel driver - I get lots of missing references.) Thanks, Fumi Okushi ___ 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] Warning Unable to allocate socket
hello i m getting follwing messages in asterisk-1.0.9 what is the reason calls are not going out. can u pls tel me how to solve this Aug 20 13:06:09 WARNING[7706]: rtp.c:829 ast_rtcp_new: Unable to allocate socket: Too many open files Aug 20 13:06:09 WARNING[7706]: channel.c:311 ast_channel_alloc: Alert pipe creation failed! Aug 20 13:06:09 WARNING[7706]: chan_sip.c:2081 sip_new: Unable to allocate channel structure Aug 20 13:06:09 NOTICE[7706]: chan_sip.c:7469 handle_request: Unable to create/find channel Aug 20 13:06:22 WARNING[7706]: acl.c:216 ast_lookup_iface: Unable to get IP of eth0: Bad file descriptor __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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] Small office setup/using analog lines w/ Asterisk
jennyw wrote: Hi, We recently tried installing Asterisk for a small office. We figured the safest way to go would be to buy from someone who sold equipment specifically for Asterisk and to use a consultant that they recommended. However ... it didn't turn out so great. Sound quality is terrible -- the echo is pretty bad, and there are popping noises, too. Callers say that people on the Asterisk end sound very faint, while people on the Asterisk end hear people maybe too loundly (might be related to the popping noises -- sounds like when you have stereo turned up too high). The reseller and the consultant both say that the most likely cause for this is using Digium cards w/ analog phone lines. Apparently, they say, sound quality can be pretty bad. I called Digium and they gave me some suggestions for settings, but nothing has worked well. So I wanted to ask others ... has anyone had good luck with using analog phone lines and Asterisk? Especially with Digium cards (we use the TDM400P)? Although from reading articles on the net it sounds like people do have a lot of echo problems, it also sounds like some people are using analog phone lines with some success. FYI, what I've mainly done is try changing echotraining, echocancel, echocancelwhenbridged, txgain, and rxgain in zapata.conf. I've heard from the reseller that what might work better is to trade the Digium cards in for VegaStream gateway. It's more expensive, but apparently has a DSP built in that should increase voice quality. Of course, they say there are no guarantees with this. They also mentioned (after the fact) that Asterisk systems don't necessarily save money. So far, the experience has been very frustrating and I'd love to hear some success stories from others (or more info on what I can realistically expect from an Asterisk system)! And, of course, some ideas on how I can get things to work better. One of the next tests will be using Asterisk with a VoIP provider to see what the sound quality is like with digital on both ends. PRI sounds like it'd be even better, but for an office w/ 5 people, it sounds pretty expensive. How do other people do this? I started using Asterisk for my own small business about a year ago. Externally we have a single analogue PSTN line (it's the house one...), an ISDN2e connection and an IAX2 connection (over 20:1 256/512kbps ADSL) with a DID in central London. The analogue line comes in to an old X100P, and the ISDN into an AVM Fritz! passive card. Internally, we have a TDM400 which talks to analogue phones in the house. In my office (which is in a different building) we have a mixture of Snom and ipDialog phones and a Grandstream ATA attached to a fax machine. We get a little echo on the ipDialog phone (but not enough to be a problem) when we talk to people on analogue phones. One of the handsets attached to the TDM400 is a DECT phone, and there's a little flurry of training noise at the beginning of an incoming call, but after that the quality is good to perfect. I'm just beginning to sell Asterisk systems. I agree that for some installations, it doesn't really make economic sense. In the UK, at least, you have to fall into a specific band of numbers-of-users and minutes-per-month for IP telephony to show a saving. Some of the small 3-line-8-extension systems from (say) Panasonic will be cheaper than Asterisk once the hardware is bought and the time (or consultancy) applied. Of course, these systems don't have much in the way of flexibility or features, and I'm talking at the moment to a company that has three sites, is using Cisco's Call Manager, and has an Asterisk system merely to convert the H.323 from the Cisco to IAX2. In this case, * could replace the CCM system in its entirety. By the time you have 100 users, * is a no-brainer in economic terms. Small users only really save (IMHO) if they a) use an awful lot of minutes (or call abroad a lot), b) need flexibility of features, or c) need internal control. Of course there may be local or exceptional circumstances which make this all a load of rubbish ! YMMV. Oh, and on echo; read : http://lists.digium.com/pipermail/asterisk-users/2005-March/096754.html jd -- John Daragon [EMAIL PROTECTED] argv[0] limited Lambs Lawn Cottage, Staple Fitzpaine, Taunton, TA3 5SL, UK v +44 (0) 1460 234068 f +44 (0) 1460 234069 m +44 (0) 7836 576127 ___ 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] ISDN BRI voice one way only
Hi, Klemens Kasemaa schrieb: hi PSTN <--> [Teles ISDN / Asterisk] <--> SIP client When call is made through ISDN, no matter if taken from PSTN or Asterisk side, person in PSTN side can hear perfectly but in Asterisk side I only hear a very scrambled or very low quality voice, words repeated several times. Same is with echo test (call taken from PSTN) Get a CAPI module for your Teles and try chan_capi-cm from http://sourceforge.net/projects/chan-capi/ Regards, Stefan -- (o_ Stefan Gofferje| SCLT //\ Reg'd Linux User #247167 | VCP #2263 V_/_ Heckler & Koch - the original point and click interface ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: [Asterisk-Dev] IM patch
harry gaillac wrote: > Hello, > > I patched asterisk cvs head sources with > http://juraj.bednar.sk/work/software/asterisk/messaging/ > and presnce patch without success. > > asterisk send "405 method not allowed" to sender. > I use polycom ip300. THat is a response to the polycom's PUBLISH request, a method that is not implemented. /O ___ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: [Asterisk-Dev] IM patch
harry gaillac wrote: > Hello, > > I patched asterisk cvs head sources with > http://juraj.bednar.sk/work/software/asterisk/messaging/ > and presnce patch without success. > > asterisk send "405 method not allowed" to sender. > I use polycom ip300. THat is a response to the polycom's PUBLISH request, a method that is not implemented. /O ___ 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] Small office setup/using analog lines w/
Jenny - I'm glad I'm not the only one! I just installed Asterisk on Friday and I spent all day trying to de-gremlin my system. I'm glad I'm doing it for myself and we haven't switched from our legacy system yet, but I have a potential client that wants to see how well I can implement this system for myself... (Speaking of which, don't forget that your supplier and consultant probably make more off of other phone systems and a lot of old-line suppliers are kind of afraid of Asterisk. There were some big companies that lost out when the State of Oregon's court system went to Asterisk for their teleconferencing needs... 600 MeetMe sessions a day does say something, I think.) First, it would help -me- to know precisely what hardware you're using (including network... switches and cards) and what phones you're using. You can email me off-list if you want. I'm personally using a AMD Sempron-based system with a TDM-04B (4x FXO) in it, routing out through an IntelPro 10/100 card. The data switch is currently a crappy D-Link, but more on that and the network card in a second. The other end is one soft phone on Linux and two SPA-841 hardware phones. When I first installed, the echo was terrible. Tweaking around with the echo settings in zapata.conf endlessly didn't do a whole lot. So, using my go-go-gadget troubleshooting instincts (and my handy-dandy go-go-gadget credit card) I figured the problem might be network latency ... so I replaced the crappy realtek network card with a high-quality Intel one, and borrwed a Dell managed switch from a client for the day. Voila, a few more tweaks to the echo learning rate and my echo problems went away, and I spent 45 minutes learning about my sister's love life in the wilds of downtown San Francisco. (Note to self: Don't use family to test telecom systems.) The clicking persisted, though. I thought it was the phone at first, but if you're experiencing it too... well, apparently it's not. I searched the archives and couldn't find any other reference to clicking noises, so I'm at a loss but I'm hoping it's wiring-based and heading back to the office tomorrow to try and solve it. I think you're right that Digium is focusing on their T-1/E-1 market. I've noticed it when I called with a question, and that really ticks me off. It might be a decent, sound business decision based on profits for the different cards, but it's leaving a lot of us smaller business operators -- those who might take great experiences with Digium and Asterisk products into potential large customers -- swinging in the wind. -Karl Katzke Streetlamp Software Solutions http://www.streetlampsoftware.com > Hi, > > We recently tried installing Asterisk for a small office. We figured the > safest way to go would be to buy from someone who sold equipment > specifically for Asterisk and to use a consultant that they > recommended. However ... it didn't turn out so great. Sound quality is > terrible -- the echo is pretty bad, and there are popping noises, too. > Callers say that people on the Asterisk end sound very faint, while > people on the Asterisk end hear people maybe too loundly (might be > related to the popping noises -- sounds like when you have stereo turned > up too high). The reseller and the consultant both say that the most > likely cause for this is using Digium cards w/ analog phone lines. > Apparently, they say, sound quality can be pretty bad. > > I called Digium and they gave me some suggestions for settings, but > nothing has worked well. So I wanted to ask others ... has anyone had > good luck with using analog phone lines and Asterisk? Especially with > Digium cards (we use the TDM400P)? Although from reading articles on the > net it sounds like people do have a lot of echo problems, it also sounds > like some people are using analog phone lines with some success. > > FYI, what I've mainly done is try changing echotraining, echocancel, > echocancelwhenbridged, txgain, and rxgain in zapata.conf. I've heard > from the reseller that what might work better is to trade the Digium > cards in for VegaStream gateway. It's more expensive, but apparently has > a DSP built in that should increase voice quality. Of course, they say > there are no guarantees with this. They also mentioned (after the fact) > that Asterisk systems don't necessarily save money. So far, the > experience has been very frustrating and I'd love to hear some success > stories from others (or more info on what I can realistically expect > from an Asterisk system)! And, of course, some ideas on how I can get > things to work better. > > One of the next tests will be using Asterisk with a VoIP provider to see > what the sound quality is like with digital on both ends. PRI sounds > like it'd be even better, but for an office w/ 5 people, it sounds > pretty expensive. How do other people do this? > > Thanks in advance for any pointers! > > Jen > > > ___ > Asterisk-Users mailing l
