Re: [Asterisk-Users] New version of DIAX (0.9.8a) available nowfor free download
Hi, - Original Message - From: Juan J. Sierralta P. [EMAIL PROTECTED] Cool. It is posible to use the GSM phone as a DIAX headset ? At least there is posible to transmit audio using Bluetooth. Unfortunately not, because the GSM phone does not support Audio Gateway profile (just Headset profile). It can connect only with the headset. ..but.. you can use the Bluetooth headset for DIAX and the GSM phone as CallerID/Dialer. .. and all this even when the computer screen is locked. Best regards, Dan ___ 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] Introduction
Greeting to all, Hello my is James Jones. I am one of the new Tech Support people at Broadvoice. I have signed up for the Asterisk mailing list to better understand some of our customer's need and to learn more about Asterisk and what it can do. I can help answer any general questions about our services. I have been a Linux user for more than 8 years now and I understand for the most part how the open source community works. I would to help by providing help when I can to the group. I would like to note any information you may receive through me from this list or my VoIP forum (http://www.outcast.ws) is not the official policy or fully supported by Broadvoice, INC. You can receive official support by contacting us at [EMAIL PROTECTED] or by calling 978-418-7300. I do this help with the progress of a tool which I find interesting and useful to the VoIP. -James --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.700 / Virus Database: 457 - Release Date: 6/6/2004 ___ 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] PC Mag Online article on Asterisk
So the kid installed it in his house and wrote an article about it. Hey, it's free publicity. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steven Critchfield Sent: Wednesday, June 09, 2004 5:39 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] PC Mag Online article on Asterisk On Wed, 2004-06-09 at 16:52, Jean-Denis Girard wrote: Very positive article on Asterisk ! http://www.pcmag.com/article2/0,1759,1607896,00.asp Maybe positive, but some of the terminology was borked such as quad-span E1 and T1 channel bank PCI cards for digital connectivity. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ 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] Sending # and Asterisk Transfer Conflict
On Wed, 2004-06-09 at 17:03 -0400, Karl J. Vesterling wrote: ANother workaround I have through of would be to use 99 for outgoing, but with no transfer options in the Dial() setting. This of course implies you're expecting to interact with a remote IVR. Last week I had to spend all day moving around M$'s IVR system, joke was I managed to avoid the premium cost number. The point is that I was using an Alcatel 4200 legacy PBX and it can not interact with IVRs unless you tap a code before the first interaction, afterwards IVR works, at the end of the call everything reverts to normal. When I used another different Alcatel a while ago, quite by accident, I hit flash, more through frustration than anything else, I found that when I got the line back I was at the next menu. If * is sensing the # couldn't it sense another code to switch to IVR mode until the end of the call, OK transfer would be disabled but that may be the least of your worries. Just my 0.02¤ worth -- Dave Cotton [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Changes in VoiceMail
There appear to have been some changes made recently to the way VoiceMail works. Previously if you pressed 7 whilst a message was playing, it would delete the message. Now if you press 7 whilst a message is playing it takes you to a menu and then you have to press 7 again to delete the message. Was this an intentional change? Simon Brown ___ 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] Sending # and Asterisk Transfer Conflict
The point is that I was using an Alcatel 4200 legacy PBX and it can not interact with IVRs unless you tap a code before the first interaction, Ugh, yes.. I remember those.. nice ring tones and stuff, the phones weren't all THAT bad to use.. but that pressing something before you can send DTMF to line thing really bugged me.. You had to press B or something.. and I think there was a timer issue too? There was something in the lists a while back, or maybe on the bug tracker, where you had to hit # twice in order to activate the transfer function. Two #'s with delays were ok, or # followed by any other digit was fine too. Maybe google for double # transfer (or wait for someone to step up and claim authorship!) Paul ___ 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] Sending # and Asterisk Transfer Conflict
The internal call parking SUCKS .. that's why tony and I wrote valetparking but nobody seems to have liked it so we gave up trying to give it away since everyone was so down on it. Hehe, I was just adding valetparking to the asterisk wiki software addons. Maybe you write a page there that describes valetparking more? ___ 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] Integration with a Siemens HiCom 150E / HiPath 3750
Hello Martin, how would you like to integrate? PRI (E1) or BRI (ISDN)? Besides of making calls with VoIP from PC to PC, we'd like that our people abroad could dial company internal extensions through Asterisk using a SIP client. On a second approach, the same people abroad could dial the PSTN using the same method... That should not affect your integration with the legacy pbx. Our scenario is: DTAG -- * HICOM PRI | PRI | SIP Please tell me the magical receipt on a step-by-step basis, as I'm not much into this telco world ;) Sorry, that is not that easy because the receipt depends much on the circumstances. What connection do you have between pstn and hicom? And you should read everything about the leagacy integration, so you will get an idea, what you want to have. Bye Felix ___ 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] Dyn Exten
Jose R. Ortiz Ubarri wrote: Hi: Is DynExtebDB module still working?? Don't bother. That application should have never been written. Jeremy McNamara ___ 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] Fax detected, but no fax extension
Hi Patrick, could you please give us a feedback if that have worked? Because I have hacked the source to disable fax.. Thanks Felix -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nicolas Gudino Sent: Wednesday, June 09, 2004 8:48 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Fax detected, but no fax extension Hi Patrick Patrick J. Conroy wrote: Hello all, I have a fax machine attached to one of the FXS ports on my channel bank running into one of the spans of my TE405P. Every time I try to send a fax, I get the error Fax detected, but no fax extension in asterisk. Does anyone know why this would happen? The only other reference I have found that relates to this in the list said to enable OLD_DSP_ROUTINES and rebuild and reinstall asterisk. I have done that, but there is no change. If you used CVS-HEAD there is a new faxdetect parameter for zapata.conf . I have not tried, but it might solve your problem. ;faxdetect=both ;faxdetect=incoming ;faxdetect=outgoing ;faxdetect=no -- Nicolas Gudino House Internet S.R.L. Buenos Aires - Argentina ___ 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] New version of DIAX (0.9.8a) available nowfor free download
Hi Dan, could you support alaw/mlaw? Is that a big problem? Regards Felix ___ 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] CDR for transfered calls
Hi, -Original Message- Yes. The issue here is that billing information is never correct in such a scenario, since the call duration on the registered asterisk machine (the one that is kicked from the path) is no longer correct. To fix this a notransfer=yes is mandatory, but that defies the practicality of having a voip conversation take the shortest path. Sure... So, this issue is sort of a bug and it really needs to be implemented then! I'm afraid its not that simple. Unless I'm misunderstanding the concepts of IAX(2) design, it does not support such behaviour _by design_. Who knows what would break if someone hacked our desires in there. A solution then would be: choose another protocol. Technically you could spin off an IAX2-cdr channel that supports it, but that would require duplicate efforts to maintain both channels. My current position is 'deal with it' and accept the extra traffic. If someone with more knowledge about IAX(2) can comment on the feasability of this behaviour we may proceed in that direction, otherwise we're just stuck. Florian ___ 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] NetworkWorld article on Open Source Telephony
Maybe, maybe not... Depending how one designs the GUI! No, I think that GUIs though needed, do limit flexibility because the information density is limited on the user-system direction (they are better on the System-user end, however). However, this is NOT an argument not to package them with the project. When I was a newbie at Samba, I used to use SWAT (Samba Web Admin Tool) all the time. However, eventially I discovered that it became easier for me to just modify the config files. This process would not have occurred as easily if I had to learn the config files at first. Guis are great at allowing less knowledgable people to administrate the server with relative competency. They are not so good at allowing one to really engineer the right solution for a customer. So I think both interface types are needed. Another example is X11 on Linux with lots of admin tools. Great for newbies, just not the best for experts. Yes, you are right!! However, GUI for newbie's will help some people to overcome the first hurdles, and then plunge into more advanced stuff! ___ 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] NetworkWorld article on Open Source Telephony
Yes, you are right!! However, GUI for newbie's will help some people to overcome the first hurdles, and then plunge into more advanced stuff! One thing quote a lot of companies do is outsource the initial configuration, because they simply don't have the technical skills initially. But what you want then is a way to easily go in and add an extension, remove someone who's left, setup hunt groups, etc, etc. It's more the general day to day maintenance that needs to be addressed, editing really complex IVR's, dialplans, etc I think should be left to the people who know what there doing. (Although there's nothing stopping adding an advanced interface too..) Just my thoughts =) Kind Regards, Chris Bond ___ 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] Primustel a.k.a. Lingo $20/month unlimited service
Hi all, I just saw this article about this new offer from Lingo.com: http://www.techweb.com/wire/story/TWB20040607S0008 $20 monthly plan with unlimited local and long-distance calling in North America (US Canada) and Western Europe. Plus first three months free and free equipment. It doesn't say what hardware they send you. Sounds like a very good deal. I searched the list and voip-wiki and couldn't find any reviews about their service. Has anyone tried them? How is the service? Does it work with *? What codec are they using? Thanks in advance for any answers, Simon in New Orleans ___ 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] AS5300 and Asterisk
Eric J Merkel wrote: Then you won't be able to us it for terminating voice. Sorry :( Eric Oh well, it was worth a though. Thanks for the answers though. I knew I should have gone for the 5350. If you get 'voice' cards for the AS5300 it'll work, and if you had gone for the 5350 you would still have to have purchased 'voice' licenses, rather than 'modem' licenses (and appropriate IOS) otherwise it still wouldnt have worked! ___ 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] New version of DIAX (0.9.8a) available nowfor free download
Hi Felix, - Original Message - From: ePyron Felix Deierlein [EMAIL PROTECTED] could you support alaw/mlaw? Is that a big problem? DIAX is based on iaxclient library which is another project. I have just done some modifications to comply with my app, but not added any new feature, like a new codecs, etc. As soon as this will be implemented in the library it will be available in DIAX too. Best regards, Dan ___ 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[2]: [Asterisk-Users] Mine strangest asterisk problem ever ....
Hello Brent, Wednesday, June 9, 2004, 7:13:52 PM, you wrote: BF On Wed, 9 Jun 2004, Alessio Focardi wrote: Asterisk with one HFC isdn card, using the zaptel driver bristuff All works ok, but voice coming in/out of the isdn card is out of sync, squelky and disrupted, UNTIL I PUT SOME LOAD TO THE PC, let say launching xwindows. BF Alessio, BF When I was having similar issues the Digium Support folks reccommended BF using hdparm. hdparm sets hard drive parameters (hence hdparm) BF You can try doing different things with it, but I know that I am currently BF set to level 3 rather than 5 as default with RedHat. I checked hdparm googling around, what parameter have you set to 3 instead of 5 ? I'm still banging my head against the wall, the only fix I found to my problem by now is running a script that put some load on the machine to have voice in sync BF - Brent BF ___ BF Asterisk-Users mailing list BF [EMAIL PROTECTED] BF http://lists.digium.com/mailman/listinfo/asterisk-users BF To UNSUBSCRIBE or update options visit: BFhttp://lists.digium.com/mailman/listinfo/asterisk-users -- Best regards, Alessiomailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] CDR for transfered calls
Sure... So, this issue is sort of a bug and it really needs to be implemented then! I'm afraid its not that simple. Unless I'm misunderstanding the concepts of IAX(2) design, it does not support such behaviour _by design_. Who knows what would break if someone hacked our desires in there. A solution then would be: choose another protocol. Technically you could spin off an IAX2-cdr channel that supports it, but that would require duplicate efforts to maintain both channels. My current position is 'deal with it' and accept the extra traffic. If someone with more knowledge about IAX(2) can comment on the feasability of this behaviour we may proceed in that direction, otherwise we're just stuck. In that case, I suppose we have to put up with it untill it gets to be a real pain in the neck :) ___ 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] SIP Registration seems to timeout
Hi. Thanks for tipping me off with the new firmware. I installed it and tested the codec. Has more delay but seems to be better quality than what I was using before. Anyways, that didn't fix the SIP Registration Failure that I am getting. Any ideas? S. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Richard Neese Sent: Wednesday, June 09, 2004 7:07 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] SIP Registration seems to timeout try changing your codec to ilbc and make sure that his gs has the latest flash to support 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
RE: [Asterisk-Users] NetworkWorld article on Open Source Telephony
Chris Bond wrote: Yes, you are right!! However, GUI for newbie's will help some people to overcome the first hurdles, and then plunge into more advanced stuff! One thing quote a lot of companies do is outsource the initial configuration, because they simply don't have the technical skills initially. But what you want then is a way to easily go in and add an extension, remove someone who's left, setup hunt groups, etc, etc. It's more the general day to day maintenance that needs to be addressed, editing really complex IVR's, dialplans, etc I think should be left to the people who know what there doing. (Although there's nothing stopping adding an advanced interface too..) It could not be said better ! :) ___ 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: Compiling Asterisk with G.723.1
Tony Hoyle wrote: Steve Underwood wrote: I didn't say one patent covered all the world. I said the patents on codecs exist all over the world. WIPO is simplifying this a bit, but its still pretty expensive to get a patent everywhere. I know of no country where the key aspects of a codec cannot be patented. Outside the US you can't patent software or algorythms, and a codec is (usually) both of these, therefore not patentable outside the US. This is what allows things like the xvid project to exist, for example, which breaks several US patents... Fraunhoffer somehow apparently managed to get some in europe but it was never decided whether they were valid or not (commonly it is thought that they'd have failed under legal challenge as the wording of EU patent law is very clear). Try looking up the EU patents related to any of the ETSI codecs, like GSM EFR, half rate, AMR, etc. If Fraunhoffer's patents can be challenged, they must have screwed up the way they worded them. === Hello, I think that the discussion has strayed from its original subject: the subject is WHERE is the library for the G723.1 codec in Asterisk. There are many people/companies/organizations who need G723.1. Although apparently it's not a problem using a patented codec like G723.1 outside of the USA, most of us would gladly pay a reasonable per-channel fee for it's usage, like in the case of the G729 which Digium offers. But since it is not available in this manner, I think it's only fair to provide the source code for compilation/usage at least outside of the US. I know that quite a few Asterisk users have compiled G723.1 in their box. Like many others, I would like to have this code and be able to compile it in my box. In fact, many of us would even pay a reasonable sum in order to have the code, if the people who already have it use it in their boxes are not willing to share for free. Regards, Randy Ackers. _ MSN 8 helps eliminate e-mail viruses. Get 2 months FREE*. http://join.msn.com/?page=features/virus ___ 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] EU on VoIP
Internet telephony (VoIP): Regulators and industry debate 'irreversible' trend The Commission is weighing up its options on how to regulate internet telephony. Major telecoms operators are already proposing services to avoid being squeezed out of the market. More at: http://www.euractiv.com/cgi-bin/cgint.exe/1?204OIDN=1507828-tt= Cheers, Philipp ___ 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] EU on VoIP
On Thu, 2004-06-10 at 11:41 +0200, Philipp von Klitzing wrote: telephony. Major telecoms operators are already proposing services to avoid being squeezed out of the market. I dont know if you can call Free/OneTel a major yet, but here in France they offer VoIP to all connected to their ADSL via a FreeBox with free calls to fixed lines in mainland France. BBC World Business News seemed to suggest BT has also seen the writing on the wall. Cheers -- Dave Cotton [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Using Asterix and Hylafax with Eicon DIVA E1
Hi, I would like to set up a high density FAX/PBX server, am looking at using Eicon Diva E1 card with Asterisk and Hylafax sharing channels, is this possible. I know Extensions can be reserved for voice OR fax with the combination of chan_capi used for * voice, capi4hylafax on fax but then channels must be dedicated rather than shared ie. would like to be able to answer voice/fax on same DID. This is high density solution 30channel FAX/Voice. Is there a solution which anybody uses which gets round this limitation. Would also like to hear any usefull comments +/- on an similar config. Br /Kev./ ___ 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] Zaphfc and BRI problems in Portugal...
I have configured several ISDN cards in Brazil in Cisco Routers. There is a configuration called compand-type (ulaw alaw) (Cisco). They are different between US and Brazil. The sound is very distorted when in the wrong configuration. The difference is between 56 bit and 64 bit ISDN. Maybe that s your problem between Portugal and England. I hope it helps. best regards, Flvio Gonalves [EMAIL PROTECTED] 06/07 8:58 am Hi - Anyone using zaphfc cards (bri-stuff-0.0.2) on a Portugal Telecom BRI service? I am getting loads of errors and the audio is very distorted all the time. System is a Dell PIII/933 MHZ with SCSI. Here in the UK we have perfect audio with a Compaq PIII/500 MHZ SCSI system. Files are identical in both systems as below: Is there anything magic about Portuguese ISDN? The PCI bus in the Portuguese system does seem to be sharing interrupts but I am not convinced this is the cause of the problem, as zttool reports no missed interrupts, and looking at /proc/interrupts seems to indicate approx 8000 per second. [Zaptel.conf] span=1,1,3,ccs,ami bchan=1-2 dchan=3 [zapata.conf] switchtype = euroisdn signalling = bri_cpe nationalprefix=0 internationalprefix=00 pridialplan=unknown echocancel=yes overlapdial=no ;callprogress=yes immediate=no group = 1 context=inboundpstn channel = 1-2 Jun 7 12:29:50 localhost kernel: zaphfc: bchan rx fifo not enough bytes to receive! (z1=7719, z2=7712) Jun 7 12:29:50 localhost kernel: zaphfc: bchan rx fifo not enough bytes to receive! (z1=8072, z2=8065) Jun 7 12:29:50 localhost kernel: zaphfc: bchan rx fifo not enough bytes to receive! (z1=1253, z2=1246) Jun 7 12:38:57 localhost kernel: zaphfc: bchan rx fifo not enough bytes to receive! (z1=8142, z2=8135) Help! Rgds Tim Robinson -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Julian Pawlowski Sent: 07 June 2004 11:10 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Updated: Advanced German Configuration Hi Folks, just updated my current Dialplan available on http://capi4linux.thepenguin.de/download/asterisk/config/ It now heavily uses ODBC database (MySQL) to hold most of the data like extensions and incoming connection numbers. An example databasefile is also included. The denylist has also been updated for future and current settings of the german telecom regulator. Again I added many comfortable features (vertical service codes). The following features are supported to be dialed from phone until now: - language setting - follow me - Call Forward Unconditional - Call Forward on No Answer - Call Forward on Busy - Individual Speed Dialing - Break Call Forward - Phone Lock - Redial last called external number - Reset Phone Settings There are much more features. You will notice them by reading the dialplan yourself. It is fairly good commented so gifted people won't have any problems. Comments and proposals appreciated. Regards, Julian Pawlowski Verschicken Sie romantische, coole und witzige Bilder per SMS! Jetzt neu bei WEB.DE FreeMail: http://freemail.web.de/?mc=021193 ___ 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] SIP Registration Failed !!(Need Help)
Hi All, I am trying to Register Asterisk PBX to a SIP Server. But SIP Server gives the following response to Asterisk: 400 Bad Request . Asterisk sends the Register Message to SIP server with the URI: sip: domain_name_sip-server. BUT URI should be of the format: sip: user@ domain_name_sip-server. I have configured Sip.conf file for registration as: user: password@ domain_name_sip-server. Can anybody suggest me, how to resolve this error. Thanks in advance Dinesh
Re: [Asterisk-Users] AS5300 and Asterisk
Dear Jimenez, You have to configure a dial-peer in Cisco box. A 2611 with a NM-HDV-E. It works. The configuration is something like: [Cisco] dial-peer voice 8000 voip protocol sipv2 codec g711 dest pattern 4... (Whatever says your dialing plan) session target ipv4:(ip address of your asteriks box) PS: I dont t remember the exact syntax for Cisco. If you have problems with commands above use ? to find the exact syntax. [Asterisk] In Asterisk configuration you can configure extensions.conf exten = _NXX,1,Dial(SIP/[EMAIL PROTECTED] cisco box ip address) Don t forget, when the call arrive in AS5300 you have to send to another dial-peer (pots if you want to send to PSTN or VOIP if ou want to send to another voip call leg). Probably you will have to create another dial-peer like below. dial-peer voice 8001 pots destination-pattern [2-9].. port 0:15 direct-inward-dial I hope it helps, Flavio Goncalves [EMAIL PROTECTED] 06/09 1:14 am Hey all, I have an as5300 I use for dial in customers, we have 4 PRIs on it. We have a few free channels on it. I'm wondering if I setup SIP on the as5300 I can have asterisk use the free channels for dial out. I'd still have to use my TDM04B for incoming calls, but at least I can expand my outgoing. Anyone done anything like this before? I've never messed with VoIP on Cisco equipment before. -- Daniel Jimenez djimenez[at]pobox[dot]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 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] New version of DIAX (0.9.8a) available nowfor free download
On 10/06/2004 at 09:04 Dan wrote: Hi, - Original Message - From: Juan J. Sierralta P. [EMAIL PROTECTED] Cool. It is posible to use the GSM phone as a DIAX headset ? At least there is posible to transmit audio using Bluetooth. Unfortunately not, because the GSM phone does not support Audio Gateway profile (just Headset profile). It can connect only with the headset. ..but.. you can use the Bluetooth headset for DIAX and the GSM phone as CallerID/Dialer. .. and all this even when the computer screen is locked. Best regards, Dan Any chance of getting this to work with Nokia phones Dan? Andy ___ 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: Compiling Asterisk with G.723.1
Randy Ackers wrote: Tony Hoyle wrote: Steve Underwood wrote: I didn't say one patent covered all the world. I said the patents on codecs exist all over the world. WIPO is simplifying this a bit, but its still pretty expensive to get a patent everywhere. I know of no country where the key aspects of a codec cannot be patented. Outside the US you can't patent software or algorythms, and a codec is (usually) both of these, therefore not patentable outside the US. This is what allows things like the xvid project to exist, for example, which breaks several US patents... Fraunhoffer somehow apparently managed to get some in europe but it was never decided whether they were valid or not (commonly it is thought that they'd have failed under legal challenge as the wording of EU patent law is very clear). Try looking up the EU patents related to any of the ETSI codecs, like GSM EFR, half rate, AMR, etc. If Fraunhoffer's patents can be challenged, they must have screwed up the way they worded them. === Hello, I think that the discussion has strayed from its original subject: the subject is WHERE is the library for the G723.1 codec in Asterisk. There are many people/companies/organizations who need G723.1. Although apparently it's not a problem using a patented codec like G723.1 outside of the USA, most of us would gladly pay a reasonable per-channel fee for it's usage, like in the case of the G729 which Digium offers. But since it is not available in this manner, I think it's only fair to provide the source code for compilation/usage at least outside of the US. I know that quite a few Asterisk users have compiled G723.1 in their box. Like many others, I would like to have this code and be able to compile it in my box. In fact, many of us would even pay a reasonable sum in order to have the code, if the people who already have it use it in their boxes are not willing to share for free. Regards, Randy Ackers. _ MSN 8 helps eliminate e-mail viruses. Get 2 months FREE*. http://join.msn.com/?page=features/virus ___ 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 I agree with Randy, G.723.1 would be extremely useful to many. And since G.723.1 could be used outside of the US from what I understand, it would be very practical if the source code was available for compilation use on Asterisk. Thanks, Vlasis Hatzistavrou. ___ 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] New version of DIAX (0.9.8a) available nowfor free download
Hi Andy, - Original Message - From: Andy Powell [EMAIL PROTECTED] Any chance of getting this to work with Nokia phones Dan? No chance unfortunately.. Nokia does not support the extended AT commands set needed to control phone keyboard and display. This is one of the reasons I like Ericsson;-) Best regards, Dan ___ 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] Sending # and Asterisk Transfer Conflict
Karl J. Vesterling wrote: It would be nice to have an option where dialing two #'s would send a single #. I too have had difficulty with this. My workaround is to use my cell when calling an IVR. ANother workaround I have through of would be to use 99 for outgoing, but with no transfer options in the Dial() setting. This of course implies you're expecting to interact with a remote IVR. PS: SIP Transfer only works with SIP phones... For those of us using Zapata channels we're pretty much screwed if that's your only option. On Zap channels we use FLASH to do a transfer. ___ 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] Please help !!!! - IAX, MYSQL - Cant make calls
Hi, I apologies for reposting this message, I am getting no where in solving this issue. And I am sure it is something very simple. I have two Firefly clients configured, If I use iax.conf to specifiy the accounts everything seems to work as expected. However If I use mysql, I can register and recieve calls on the firefly accounts (from SIP etc) but can not make calls between the two or anything else. I get a message on firefly Call ended with xxx reason : no authority found On the console I see the following message CHAN_IAX2 ... Socket_Read: Rejected connect attempt from IP Please help, Umar. Yahoo! Messenger - Communicate instantly...Ping your friends today! Download Messenger Now http://uk.messenger.yahoo.com/download/index.html ___ 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] Changes in VoiceMail
Simon Brown [EMAIL PROTECTED] wrote:
There appear to have been some changes made recently to the way VoiceMail
works. Previously if you pressed 7 whilst a message was playing, it would
delete the message. Now if you press 7 whilst a message is playing it
takes you to a menu and then you have to press 7 again to delete the
message.
Was this an intentional change?
It appears to be a bug. The following patch fixes it for me:
*** app.c 1 Jun 2004 19:38:06 - 1.21
--- app.c 10 Jun 2004 11:44:11 -
***
*** 473,479
break;
if (stop strchr(stop, res)) {
- res = 0;
break;
}
}
--- 473,478
--
_/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/
_/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h
_/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED]
_/ _/ _/_/_/_/ _/_/_/_/ _/_/
___
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Re: [Asterisk-Users] Re: Compiling Asterisk with G.723.1
So simple question, without googling: Where can the g723.1 and g723.1a be found... so a 'EU-patch' can be make. I'm able to host it in Amsterdam. Greetings, Stefan de Konink On Thu, 10 Jun 2004, Vlasis Hatzistavrou wrote: Randy Ackers wrote: Tony Hoyle wrote: Steve Underwood wrote: I didn't say one patent covered all the world. I said the patents on codecs exist all over the world. WIPO is simplifying this a bit, but its still pretty expensive to get a patent everywhere. I know of no country where the key aspects of a codec cannot be patented. Outside the US you can't patent software or algorythms, and a codec is (usually) both of these, therefore not patentable outside the US. This is what allows things like the xvid project to exist, for example, which breaks several US patents... Fraunhoffer somehow apparently managed to get some in europe but it was never decided whether they were valid or not (commonly it is thought that they'd have failed under legal challenge as the wording of EU patent law is very clear). Try looking up the EU patents related to any of the ETSI codecs, like GSM EFR, half rate, AMR, etc. If Fraunhoffer's patents can be challenged, they must have screwed up the way they worded them. === Hello, I think that the discussion has strayed from its original subject: the subject is WHERE is the library for the G723.1 codec in Asterisk. There are many people/companies/organizations who need G723.1. Although apparently it's not a problem using a patented codec like G723.1 outside of the USA, most of us would gladly pay a reasonable per-channel fee for it's usage, like in the case of the G729 which Digium offers. But since it is not available in this manner, I think it's only fair to provide the source code for compilation/usage at least outside of the US. I know that quite a few Asterisk users have compiled G723.1 in their box. Like many others, I would like to have this code and be able to compile it in my box. In fact, many of us would even pay a reasonable sum in order to have the code, if the people who already have it use it in their boxes are not willing to share for free. Regards, Randy Ackers. _ MSN 8 helps eliminate e-mail viruses. Get 2 months FREE*. http://join.msn.com/?page=features/virus ___ 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 I agree with Randy, G.723.1 would be extremely useful to many. And since G.723.1 could be used outside of the US from what I understand, it would be very practical if the source code was available for compilation use on Asterisk. Thanks, Vlasis Hatzistavrou. ___ 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] New version of DIAX (0.9.8a) available nowfor free download
Hi Dan On 10/06/2004 at 14:01 Dan wrote: Hi Andy, - Original Message - From: Andy Powell [EMAIL PROTECTED] Any chance of getting this to work with Nokia phones Dan? No chance unfortunately.. Nokia does not support the extended AT commands set needed to control phone keyboard and display. This is one of the reasons I like Ericsson;-) Best regards, Dan Ok, but can I still used my BT headset and a BT dongle on the PC to speak? I'm thinking it's a bit easier to carry the headset about for answering calls. For dialing I'm happy to pick up a proper phone or if I'm at the PC just use the DIAX interface. Possible? Thanks Andy ___ 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] Sending # and Asterisk Transfer Conflict
On Thursday 10 June 2004 03:03, Holger Schurig wrote: Maybe you write a page there that describes valetparking more? Yes, please! ValetParking is supposed to do practically everything yet there is next to no documentation on how to make it do _anything_ -- Let's get some killer documentation on this app so that there's no choice BUT to put it into Asterisk itself. No I'm not being sarcastic -- as I said in a previous email it took Brian 3 attempts to get the BASIC functionality of app_valetparking and the differences between it and normal parking through my thick skull -- We Need (More) Documentation on this thing and since there are only two people who really understand what it is capable of... well those people need to finish up the app by including some decent documentation and examples! -A. ___ 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: Compiling Asterisk with G.723.1
Hello, I have the whole library (G.723.1 and G.723.1b) downloaded from ITU, but it doesn't compile with Asterisk out-of-the-box. So, unless someone else can provide a library which compiles with *, we'll have to tinker with the ITU source code (if it is possible at all). Best regards, Vlasis Hatzistavrou. Stefan de Konink wrote: So simple question, without googling: Where can the g723.1 and g723.1a be found... so a 'EU-patch' can be make. I'm able to host it in Amsterdam. Greetings, Stefan de Konink On Thu, 10 Jun 2004, Vlasis Hatzistavrou wrote: Randy Ackers wrote: Tony Hoyle wrote: Steve Underwood wrote: I didn't say one patent covered all the world. I said the patents on codecs exist all over the world. WIPO is simplifying this a bit, but its still pretty expensive to get a patent everywhere. I know of no country where the key aspects of a codec cannot be patented. Outside the US you can't patent software or algorythms, and a codec is (usually) both of these, therefore not patentable outside the US. This is what allows things like the xvid project to exist, for example, which breaks several US patents... Fraunhoffer somehow apparently managed to get some in europe but it was never decided whether they were valid or not (commonly it is thought that they'd have failed under legal challenge as the wording of EU patent law is very clear). Try looking up the EU patents related to any of the ETSI codecs, like GSM EFR, half rate, AMR, etc. If Fraunhoffer's patents can be challenged, they must have screwed up the way they worded them. === Hello, I think that the discussion has strayed from its original subject: the subject is WHERE is the library for the G723.1 codec in Asterisk. There are many people/companies/organizations who need G723.1. Although apparently it's not a problem using a patented codec like G723.1 outside of the USA, most of us would gladly pay a reasonable per-channel fee for it's usage, like in the case of the G729 which Digium offers. But since it is not available in this manner, I think it's only fair to provide the source code for compilation/usage at least outside of the US. I know that quite a few Asterisk users have compiled G723.1 in their box. Like many others, I would like to have this code and be able to compile it in my box. In fact, many of us would even pay a reasonable sum in order to have the code, if the people who already have it use it in their boxes are not willing to share for free. Regards, Randy Ackers. _ MSN 8 helps eliminate e-mail viruses. Get 2 months FREE*. http://join.msn.com/?page=features/virus ___ 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 I agree with Randy, G.723.1 would be extremely useful to many. And since G.723.1 could be used outside of the US from what I understand, it would be very practical if the source code was available for compilation use on Asterisk. Thanks, Vlasis Hatzistavrou. ___ 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] IAX Binding to 2 nic's for trunking two asterisk servers
I have a problem in that when you use IAX2 for trunking and have 2 nics one is used to connect directly to 2nd Asterisk server how do we get the outside Nic card to take IAX connections? Is there any way to get this working via two paths? There is only one bindipaddr=10.1.1.1 for internal trunk but outside address section? - \ \\_ Ariel Batista // / Avionica, Inc. -- [EMAIL PROTECTED] Ph: 786-544-1114 Fx: 305-574-0212 ___ 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: Re: DNS SRV records
Randy Bush wrote: Time for Duane to start implementing DNS SRV, since it's from now on is turned on by default in CVS head. Unless you're planning on breaking other standards my A records will keep on working just fine :) except you (likely to be ex-) customers will have problems reaching more and more of the universe. as the idiom goes, not a problem to me. randy Come on Randy. I was hoping to see the ever popular and appropriate, I encourage my competition to do exactly what you are doing... or however it is that you used to phrase that. ;) John ___ 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] Primustel a.k.a. Lingo $20/month unlimited service
They don't provide soft accounts. You need to use their D-Link box which connects back to them using MGCP. Overall service is reasonable, acceptable for home users but definitely not good enough for business use. I am just about to send their units back. Thanks, Wojtek - Original Message - From: Stephan Wik [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, June 10, 2004 4:46 AM Subject: Re: [Asterisk-Users] Primustel a.k.a. Lingo $20/month unlimited service On 10 Jun 2004, at 09:53, Simon Dorfman wrote: $20 monthly plan with unlimited local and long-distance calling in North America (US Canada) and Western Europe. Plus first three months free and free equipment. It doesn't say what hardware they send you. Sounds like a very good deal. I searched the list and voip-wiki and couldn't find any reviews about their service. Has anyone tried them? How is the service? Does it work with *? I just spoke with their tech support who says you have to use their 'hardware' to connect. He had no idea what I was talking about when I mentioned IAX or SIP :-( Stephan ___ 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] FWIW- Cisco 1750 dropped packets and choppy audio
This email is intended to document an issue for anyone searching the archives. We had a problem yesterday with _all_ iax2 and sip sessions; no reasonable conversation could be established due to extremely choppy audio in one direction only (outbound from * to distant sip phones and distant * boxes). We were running HEAD from June 8th. While diagnosing the root cause, we monitored bandwidth utilization at the asterisk-connected managed-switch as well as at the Cisco 1750 Internet interface. We observed consistent/even data flows to/from the * box, however the outbound Cisco interface indicated more inbound traffic than outbound traffic by a considerable/noticeable amount. Both iax2 and sip sessions were impacted exactly the same regardless of the codec being used. In the haste to identify the root-cause, the Cisco 1750 was rebooted (Version 12.2(4)T7) and the problem disappeared. A Service-Policy had been applied to the outbound interface for QoS purposes. Removing the policy while a poor quality session was in progress had zero impact. Unfortunitly, no other Cisco data was gathered before the reboot. We're waiting for reoccurrence to gather additional doc. We are 100% confident this is a Cisco issue as opposed to * or any other resource. (Someone, maybe Eric, mentioned a Cisco QoS bug previously on this list. Indications are this might be the bug that person had mentioned.) The Cisco had been in use for a couple of years and we've never seen this issue arise prior to yesterday. The * box had been in semi-production since late last year and has been stable (given the expected issues associated with using HEAD as opposed to Stable). There were no logged messages from the Cisco even though syslog messages are normally monitored closely. Rich ___ 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] Iax2 ringtone problem
Hi, i have a problem with iax2 and ringtone. Here is the call path pstn - asterisk - iax - firefly or any iax phone. My problem is when i receive a call on my iax phone, the ring sound is very distort and bad. If i open my sip phone, and receive a call from my pstn, the ring is like dring dring, very normal. Otherwise, it is like a machine gun with iax Help would be really appreciate on how i can fix my iax issue JF Thank ___ 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] Iax2 ringtone problem
Hi, i have a problem with iax2 and ringtone. Here is the call path pstn - asterisk - iax - firefly or any iax phone. My problem is when i receive a call on my iax phone, the ring sound is very distort and bad. If i open my sip phone, and receive a call from my pstn, the ring is like dring dring, very normal. Otherwise, it is like a machine gun with iax Help would be really appreciate on how i can fix my iax issue JF Thank ___ 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] Sending # and Asterisk Transfer Conflict
FWIW, I like the valetparking feature but the documentation sucks rocks -- you had to describe it at least three times before I *started* to understand its utility and features above and beyond normal parking. Perhaps the problem isn't so much that the people were down on it as that they didn't understand all of its abilities nor how to use them... The docs on my site were fine... they explained and gave examples of how to use it. bkw ___ 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] Sending # and Asterisk Transfer Conflict
I replaced the .c file with a note read the .c file. :) bkw -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kevin Sent: Wednesday, June 09, 2004 9:32 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Sending # and Asterisk Transfer Conflict I was excited to try this out, but I think Im stuck when I run the astxs -install apps/app_valetparking.c command. I get this error: any suggestions [EMAIL PROTECTED] asterisk]# astxs -install apps/app_valetparking.c gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\CVS-HEAD-05/22/04-12:16:21\ -DINSTALL_PREFIX=\\ -DASTETCDIR=\/etc/asterisk\ -DASTLIBDIR=\/usr/lib/asterisk\ -DASTVARLIBDIR=\/var/lib/asterisk\ -DASTVARRUNDIR=\/var/run\ -DASTSPOOLDIR=\/var/spool/asterisk\ -DASTLOGDIR=\/var/log/asterisk\ -DASTCONFPATH=\/etc/asterisk/asterisk.conf\ -DASTMODDIR=\/usr/lib/asterisk/modules\ -DASTAGIDIR=\/var/lib/asterisk/agi-bin\ -DBUSYDETECT_MARTIN -DNEW_PRI_HANGUP -fPIC -c apps/app_valetparking.c -o apps/app_valetparking.o apps/app_valetparking.c:1: parse error before gave [EMAIL PROTECTED] asterisk]# cd app - -Original Message- From: brian k. west [mailto:[EMAIL PROTECTED] Sent: Wednesday, June 09, 2004 8:42 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Sending # and Asterisk Transfer Conflict Valetparking with some creativeness would do campop.. hell you can do campon with just extension logic. bkw - Original Message - From: Karl J. Vesterling To: [EMAIL PROTECTED] Sent: Wednesday, June 09, 2004 4:58 PM Subject: RE: [Asterisk-Users] Sending # and Asterisk Transfer Conflict Yes! Camp-On would be a phreaking phantastic pheature! At 05:55 PM 6/9/2004, you wrote: Didn't know everyone was down on it. It's just not a very used feature in my office environment. What's needed is a true camp-on. That's used lots at everywhere I've ever worked, and asterisk is missing it. It has an anemic call pickup that doesn't do much for us. (or even work at the moment) Your efforts are not in vain... Nik -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of brian Sent: Wednesday, June 09, 2004 4:36 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Sending # and Asterisk Transfer Conflict The internal call parking SUCKS .. that's why tony and I wrote valetparking but nobody seems to have liked it so we gave up trying to give it away since everyone was so down on it. bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Stephen Rosebush Sent: Wednesday, June 09, 2004 2:29 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Sending # and Asterisk Transfer Conflict I have a Grandstream ATA286 and still can not find a way of issuing '#' to anything with call parking enabled.. I use call parking quite frequently and on my ATA device I can not issue a # to anything I encounter that might require it. When I push flash on my ATA device it does what it should, It puts the call I was currently in on hold so I can answer an incoming call / make another outgoing call... Same as a landline phone... I can NOT transfer using FLASH. Steve Eric Wieling wrote: What I don't understand is why people think that FLASH on a SIP ATA-like device is NOT a SIP transfer. Weird. On Wed, 2004-06-09 at 13:09, brian wrote: Yet again.. *SMACK* yes it does. bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Stephen Rosebush Sent: Wednesday, June 09, 2004 12:47 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Sending # and Asterisk Transfer Conflict Wouldn't work on an ATA device brian wrote: *SMACK* no you don't just use the native sip transfer to park it. :) bkw ___ 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 Best Regards, Karl J. Vesterling E-Mail: [EMAIL PROTECTED] Yahoo Messenger: karl_vesterling ICQ: 1548052 AOL Instant Messenger: n2vqm
[Asterisk-Users] isdn4linux and NT mode
Dear friends, I have an HFC ISDN card that I have set up in NT mode. Will this be enough to connect to an ISDN Pbx and pretend to be and ISDN line ? Tnx for any help ? -- Best regards, Alessio mailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sending # and Asterisk Transfer Conflict
On Thursday 10 June 2004 09:50, brian wrote: The docs on my site were fine... they explained and gave examples of how to use it. Hmm I did not see them. Are they still up? I saw you said you took down the .c, so I am assuming they're gone too? -A. ___ 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: Compiling Asterisk with G.723.1
The reference code does not pack or unpack the bits. It needs additional work to make a usable codec. This is true of most reference codec implementations. The bit packing arrangements depend on the application of the codec, so they are often not specified as part of the codec. Regards, Steve Vlasis Hatzistavrou wrote: Hello, I have the whole library (G.723.1 and G.723.1b) downloaded from ITU, but it doesn't compile with Asterisk out-of-the-box. So, unless someone else can provide a library which compiles with *, we'll have to tinker with the ITU source code (if it is possible at all). Best regards, Vlasis Hatzistavrou. ___ 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: Compiling Asterisk with G.723.1
Randy Ackers wrote: Tony Hoyle wrote: Steve Underwood wrote: I didn't say one patent covered all the world. I said the patents on codecs exist all over the world. WIPO is simplifying this a bit, but its still pretty expensive to get a patent everywhere. I know of no country where the key aspects of a codec cannot be patented. Outside the US you can't patent software or algorythms, and a codec is (usually) both of these, therefore not patentable outside the US. This is what allows things like the xvid project to exist, for example, which breaks several US patents... Fraunhoffer somehow apparently managed to get some in europe but it was never decided whether they were valid or not (commonly it is thought that they'd have failed under legal challenge as the wording of EU patent law is very clear). Try looking up the EU patents related to any of the ETSI codecs, like GSM EFR, half rate, AMR, etc. If Fraunhoffer's patents can be challenged, they must have screwed up the way they worded them. === Hello, I think that the discussion has strayed from its original subject: the subject is WHERE is the library for the G723.1 codec in Asterisk. There are many people/companies/organizations who need G723.1. Although apparently it's not a problem using a patented codec like G723.1 outside of the USA, most of us would gladly pay a reasonable per-channel fee for it's usage, like in the case of the G729 which Digium offers. But since it is not available in this manner, I think it's only fair to provide the source code for compilation/usage at least outside of the US. I know that quite a few Asterisk users have compiled G723.1 in their box. Like many others, I would like to have this code and be able to compile it in my box. In fact, many of us would even pay a reasonable sum in order to have the code, if the people who already have it use it in their boxes are not willing to share for free. Regards, Randy Ackers. We are hardly straying from the topic. The problem is it *not* legal to use this thing in the EU, or most other places, regardless of people trying to twist things around so they can say it is. Regards, Steve ___ 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] FWIW- Cisco 1750 dropped packets and choppy audio
On Thu, 2004-06-10 at 23:27, Rich Adamson wrote: This email is intended to document an issue for anyone searching the archives. We had a problem yesterday with _all_ iax2 and sip sessions; no reasonable conversation could be established due to extremely choppy audio in one direction only (outbound from * to distant sip phones and distant * boxes). We were running HEAD from June 8th. While diagnosing the root cause, we monitored bandwidth utilization at the asterisk-connected managed-switch as well as at the Cisco 1750 Internet interface. We observed consistent/even data flows to/from the * box, however the outbound Cisco interface indicated more inbound traffic than outbound traffic by a considerable/noticeable amount. Both iax2 and sip sessions were impacted exactly the same regardless of the codec being used. In the haste to identify the root-cause, the Cisco 1750 was rebooted (Version 12.2(4)T7) and the problem disappeared. A Service-Policy had been applied to the outbound interface for QoS purposes. Removing the policy while a poor quality session was in progress had zero impact. Unfortunitly, no other Cisco data was gathered before the reboot. We're waiting for reoccurrence to gather additional doc. We are 100% confident this is a Cisco issue as opposed to * or any other resource. (Someone, maybe Eric, mentioned a Cisco QoS bug previously on this list. Indications are this might be the bug that person had mentioned.) The Cisco had been in use for a couple of years and we've never seen this issue arise prior to yesterday. The * box had been in semi-production since late last year and has been stable (given the expected issues associated with using HEAD as opposed to Stable). There were no logged messages from the Cisco even though syslog messages are normally monitored closely. This may or may not be related, I have a Cisco 8xx (something, ADSL router) and had very poor audio in both directions, I was seeing very large number of packets/sec in both directions. I solved it by turning off all the debugging on the cisco. (I had debug ip packet turned on for packets matching a specific access-list). As a side note, I first noticed the CPU load on the cisco sitting on 100% during an attempted phone call. Re-booting your router would have also disabled all debugging, but that may not have been your issue, it is difficult to say. Hope this helps someone else go DoH! like I did :) Regards, Adam ___ 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] Sending # and Asterisk Transfer Conflict
I've been avoiding commenting on this thread because I haven't studied the code enough, or my current problem, but anyway, here is my 0.02c worth... I found the documentation to be OK, and the app seems to do some fantastic things, which the current call parking can't do. However, the real reason I was bothered to look at it was because I need to park a call from the manager interface. I almost get what I need from: exten = 799,1,ParkAndAnnounce(PARKED, 240, Zap/127-1, desks) except I don't really want to specify a channel to announce where it is parked, I want the manager interface to see where it has been parked so that it can be pulled back when needed. I also tried: exten = 800,1,ValetParkCall(auto,mylot,360,s,10,remote) but again had various problems with it... probably related to where it will get sent back to (ie, back to the operator instead of the original channel). I would like to (unless there is a better way) send a manager command to say park the call on channel and hangup the other channel, in the process tell the manager interface where the call has been parked. Currently I do this with the Action: Redirect to point it to one of the above extensions. Later I can send a Action: Redirect to redirect the parked call to the extension I want to... So, if anyone has any comments/suggestions on the above, I would appreciate it. Otherwise tomorrow I might be forced to look at the source code in more detail. This is dangerous, since in reality I don't know c ... Regards, Adam ___ 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: Compiling Asterisk with G.723.1
http://asterisk.gnuinter.net/files/digium/asterisk-ng/db1-ast/ I 'stole' the sources from them, compiled (and working) after uncommenting the makefile. Tobad only the Budgettones work with the codec and not the Cisco's :( Stefan On Thu, 10 Jun 2004, Vlasis Hatzistavrou wrote: Hello, I have the whole library (G.723.1 and G.723.1b) downloaded from ITU, but it doesn't compile with Asterisk out-of-the-box. So, unless someone else can provide a library which compiles with *, we'll have to tinker with the ITU source code (if it is possible at all). Best regards, Vlasis Hatzistavrou. Stefan de Konink wrote: So simple question, without googling: Where can the g723.1 and g723.1a be found... so a 'EU-patch' can be make. I'm able to host it in Amsterdam. Greetings, Stefan de Konink On Thu, 10 Jun 2004, Vlasis Hatzistavrou wrote: Randy Ackers wrote: Tony Hoyle wrote: Steve Underwood wrote: I didn't say one patent covered all the world. I said the patents on codecs exist all over the world. WIPO is simplifying this a bit, but its still pretty expensive to get a patent everywhere. I know of no country where the key aspects of a codec cannot be patented. Outside the US you can't patent software or algorythms, and a codec is (usually) both of these, therefore not patentable outside the US. This is what allows things like the xvid project to exist, for example, which breaks several US patents... Fraunhoffer somehow apparently managed to get some in europe but it was never decided whether they were valid or not (commonly it is thought that they'd have failed under legal challenge as the wording of EU patent law is very clear). Try looking up the EU patents related to any of the ETSI codecs, like GSM EFR, half rate, AMR, etc. If Fraunhoffer's patents can be challenged, they must have screwed up the way they worded them. === Hello, I think that the discussion has strayed from its original subject: the subject is WHERE is the library for the G723.1 codec in Asterisk. There are many people/companies/organizations who need G723.1. Although apparently it's not a problem using a patented codec like G723.1 outside of the USA, most of us would gladly pay a reasonable per-channel fee for it's usage, like in the case of the G729 which Digium offers. But since it is not available in this manner, I think it's only fair to provide the source code for compilation/usage at least outside of the US. I know that quite a few Asterisk users have compiled G723.1 in their box. Like many others, I would like to have this code and be able to compile it in my box. In fact, many of us would even pay a reasonable sum in order to have the code, if the people who already have it use it in their boxes are not willing to share for free. Regards, Randy Ackers. _ MSN 8 helps eliminate e-mail viruses. Get 2 months FREE*. http://join.msn.com/?page=features/virus ___ 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 I agree with Randy, G.723.1 would be extremely useful to many. And since G.723.1 could be used outside of the US from what I understand, it would be very practical if the source code was available for compilation use on Asterisk. Thanks, Vlasis Hatzistavrou. ___ 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 mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] GSM to ISDN or TAPI
Hi I am in the UK and am looking for a device that will allow me to connect two sim cards (read wireless lines) to either the port on the back of my fritz card or any other connection direct to the PC that provides a usable telephony interface. I will even plug two devices into a windows box and have that do ISDN to ISDN if required. All I want is two GSM lines that look like voice modems to the PC and provide full telephony interface, that is DTMF both ways CLI and a few other bits and pieces. I am looking to using asterisk as a remote IVR for looking after some equipment, but land lines are a problem. Any help is much appreciated Regards Chris. ___ 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 on Apple PPC with YDL
Fyi, Successfully compiled Asterisk on an Apple G4 PPC with Yellow Dog Linux - without any source modifications. Worked fast and smooth. - Darren ___ 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] Sending # and Asterisk Transfer Conflict
http://www.bkw.org/archives/000291.html the docs are still up. bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith Sent: Thursday, June 10, 2004 9:04 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Sending # and Asterisk Transfer Conflict On Thursday 10 June 2004 09:50, brian wrote: The docs on my site were fine... they explained and gave examples of how to use it. Hmm I did not see them. Are they still up? I saw you said you took down the .c, so I am assuming they're gone too? -A. ___ 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] GSM to ISDN or TAPI
Hi Chris, CL All I want is two GSM lines that look like voice modems to CL the PC and provide full telephony interface, that is DTMF CL both ways CLI and a few other bits and pieces. We use the Nokia 22: http://www.nokia.com/nokia/0,,56024,00.html They have worked well providing both telephony applications on remote sites and SMS support for Broadcast work in the UK (serial AT command interface). If you don't mind single band (900 or 1800 MHz GSM) operation there is an older device (Nokia Premicell) that can be sourced cheaply from eBay: http://www.nokia.com/cda1/0,1080,2700,00.html HTH Darren -- Comgate TelcoInternetBroadcast -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Chris Lee Sent: 10 June 2004 15:50 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] GSM to ISDN or TAPI Hi I am in the UK and am looking for a device that will allow me to connect two sim cards (read wireless lines) to either the port on the back of my fritz card or any other connection direct to the PC that provides a usable telephony interface. I will even plug two devices into a windows box and have that do ISDN to ISDN if required. All I want is two GSM lines that look like voice modems to the PC and provide full telephony interface, that is DTMF both ways CLI and a few other bits and pieces. I am looking to using asterisk as a remote IVR for looking after some equipment, but land lines are a problem. Any help is much appreciated Regards Chris. ___ 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] Dyn Exten
Why??? Is there another way to do Dynamic Extensions??? -- JO Jeremy McNamara wrote: Jose R. Ortiz Ubarri wrote: Hi: Is DynExtebDB module still working?? Don't bother. That application should have never been written. Jeremy McNamara ___ 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] GSM to ISDN or TAPI
Chris Lee [EMAIL PROTECTED] wrote: [...] I am in the UK and am looking for a device that will allow me to connect two sim cards (read wireless lines) to either the port on the back of my fritz card or any other connection direct to the PC that provides a usable telephony interface. My initial thoughs are to use an X100P plugged into a Premicell, as it's nice and simple, and it would clearly work well with Asterisk. The downside is that it's an analogue connection, of course. -- I want to know how God created this world. I am not interested in this or that phenomenon, in the spectrum of this or that element. I want to know His thoughts; the rest are details. - Albert Einstein ___ 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] FW: question about prepaid app_prepaid
Hi, I have compiled and installed app_prepaid module. But have problem when connect to postgres database. I guess so because after key in card number, it always play prepaid-no-aaa voice file. Anyone succeeded in configuring the app_prepaid for prepaid calling service for asterisk? Please help. Ps: where can I view the log file for this module. Thanks. Tom
[Asterisk-Users] Automating calls
Hello I have heard that i can put a file in a certain directory to get * to initiate a call. Is this true ? if so where would i look ? Best Regards Simon Garvey ___ 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] Sending # and Asterisk Transfer Conflict
Thanks, I figured it out and replaced it with the proper file. -Original Message- From: brian [mailto:[EMAIL PROTECTED] Sent: Thursday, June 10, 2004 9:51 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Sending # and Asterisk Transfer Conflict I replaced the .c file with a note read the .c file. :) bkw -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kevin Sent: Wednesday, June 09, 2004 9:32 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Sending # and Asterisk Transfer Conflict I was excited to try this out, but I think Im stuck when I run the astxs -install apps/app_valetparking.c command. I get this error: any suggestions [EMAIL PROTECTED] asterisk]# astxs -install apps/app_valetparking.c gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\CVS-HEAD-05/22/04-12:16:21\ -DINSTALL_PREFIX=\\ -DASTETCDIR=\/etc/asterisk\ -DASTLIBDIR=\/usr/lib/asterisk\ -DASTVARLIBDIR=\/var/lib/asterisk\ -DASTVARRUNDIR=\/var/run\ -DASTSPOOLDIR=\/var/spool/asterisk\ -DASTLOGDIR=\/var/log/asterisk\ -DASTCONFPATH=\/etc/asterisk/asterisk.conf\ -DASTMODDIR=\/usr/lib/asterisk/modules\ -DASTAGIDIR=\/var/lib/asterisk/agi-bin\ -DBUSYDETECT_MARTIN -DNEW_PRI_HANGUP -fPIC -c apps/app_valetparking.c -o apps/app_valetparking.o apps/app_valetparking.c:1: parse error before gave [EMAIL PROTECTED] asterisk]# cd app - -Original Message- From: brian k. west [mailto:[EMAIL PROTECTED] Sent: Wednesday, June 09, 2004 8:42 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Sending # and Asterisk Transfer Conflict Valetparking with some creativeness would do campop.. hell you can do campon with just extension logic. bkw - Original Message - From: Karl J. Vesterling To: [EMAIL PROTECTED] Sent: Wednesday, June 09, 2004 4:58 PM Subject: RE: [Asterisk-Users] Sending # and Asterisk Transfer Conflict Yes! Camp-On would be a phreaking phantastic pheature! At 05:55 PM 6/9/2004, you wrote: Didn't know everyone was down on it. It's just not a very used feature in my office environment. What's needed is a true camp-on. That's used lots at everywhere I've ever worked, and asterisk is missing it. It has an anemic call pickup that doesn't do much for us. (or even work at the moment) Your efforts are not in vain... Nik -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of brian Sent: Wednesday, June 09, 2004 4:36 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Sending # and Asterisk Transfer Conflict The internal call parking SUCKS .. that's why tony and I wrote valetparking but nobody seems to have liked it so we gave up trying to give it away since everyone was so down on it. bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Stephen Rosebush Sent: Wednesday, June 09, 2004 2:29 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Sending # and Asterisk Transfer Conflict I have a Grandstream ATA286 and still can not find a way of issuing '#' to anything with call parking enabled.. I use call parking quite frequently and on my ATA device I can not issue a # to anything I encounter that might require it. When I push flash on my ATA device it does what it should, It puts the call I was currently in on hold so I can answer an incoming call / make another outgoing call... Same as a landline phone... I can NOT transfer using FLASH. Steve Eric Wieling wrote: What I don't understand is why people think that FLASH on a SIP ATA-like device is NOT a SIP transfer. Weird. On Wed, 2004-06-09 at 13:09, brian wrote: Yet again.. *SMACK* yes it does. bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Stephen Rosebush Sent: Wednesday, June 09, 2004 12:47 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Sending # and Asterisk Transfer Conflict Wouldn't work on an ATA device brian wrote: *SMACK* no you don't just use the native sip transfer to park it. :) bkw ___ 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]
[Asterisk-Users] Cisco 7960 Tones
Hi all Does anybody know if it is possible to change the tones on a 7960 ? I guess there must be some way to edit the dial/busy/congestion tones ? /Mike ___ 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] I can't get iaxComm to connect to guest@misery.digium.com
On advice from others I dropped gnophone in favor of iaxComm. I am operating on an IBM T30 laptop Redhat Linux 2.4.20-8 with an Intel i810 audio chipset (comes in the laptop). I am using the Gnome desktop. There is no reference to alsa or oss to be found. All audio components function fine. Nothing else is running and I have an active broadband internet connection. I can ping www.digium.com but NOT misery.digium.com which may explain the retransmissions. I downloaded iaxcomm-lin-current.tar Untarred it. And did the following: [EMAIL PROTECTED] iaxcomm]# ls iaxcomm QUICKSTART README ring.raw [EMAIL PROTECTED] iaxcomm]# ./iaxcomm Gdk-CRITICAL **: file gdkgc.c: line 689 (gdk_gc_set_clip_rectangle): assertion `gc != NULL' failed. Gdk-CRITICAL **: file gdkdraw.c: line 90 (gdk_draw_rectangle): assertion `gc != NULL' failed. Gdk-CRITICAL **: file gdkgc.c: line 689 (gdk_gc_set_clip_rectangle): assertion `gc != NULL' failed. = Here I cancelled the Account window Pa_SetupDeviceFormat: HW does not support 2 channels Here the headset and microphone clicked on and I could hear what I spoke into the microphone in the headset I then typed [EMAIL PROTECTED]/s in the field above Dial and clicked Dial I received the following and heard nothing. I then Hungup and closed the window Pa_SetupDeviceFormat: HW does not support 2 channels Scheduling retransmission 9 Scheduling retransmission 9 Scheduling retransmission 8 Scheduling retransmission 8 Scheduling retransmission 7 Scheduling retransmission 7 Scheduling retransmission 6 Scheduling retransmission 6 Scheduling retransmission 9 Scheduling retransmission 8 Scheduling retransmission 7 Scheduling retransmission 6 Scheduling retransmission 5 Scheduling retransmission 5 Scheduling retransmission 9 Scheduling retransmission 8 Scheduling retransmission 7 Scheduling retransmission 5 Scheduling retransmission 6 Scheduling retransmission 4 Scheduling retransmission 4 Scheduling retransmission 9 Scheduling retransmission 8 Scheduling retransmission 7 [EMAIL PROTECTED] iaxcomm]# ___ 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 * not detecting hangup on FXO and VM going into an infinite loop
Hi everybody... I'm having an odd problem with voice mail on a recent CVS of * where it appears not to detect a hangup on FXO and * will keep treating the call as new and continue leaving voicemails until the max has been reached. It will then continue trying to leave voice mails and basically makes the system unavailble to any further incoming or outgoing calls on that FXO..has anybody seen this and if so how do I fix it? I've looked around on google and the list archives and it appears that there are others with similar problems with most people believing it to be a configuration problem. Since I don't see any bugs that have been formally posted with this description I think it most likely is...can anybody help me determine which option would be causing this behavior? I assume its in zapata.conf? Thanks! Chris -- It's easy to sit there and say you'd like to have more money. And I guess that's what I like about it. It's easy. Just sitting there, rocking back and forth, wanting that money. http://ccicolorado.org Exceptional Dogs for Exceptional People - Help Out Today! ___ 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] Iax2 ringtone problem
Hi Jean, It seems that no one on the list is interested in IAX, I have posted a couple of basic questions but no ones seems to want to answer. I guess everyone is busy right now. Anyway back to your question. When you say the ringtone , do you mean the rinback tone (what the caller hears) or bell to notify the callee that there is a call. Umar. --- Jean-Francois Dubé [EMAIL PROTECTED] wrote: Hi, i have a problem with iax2 and ringtone. Here is the call path pstn - asterisk - iax - firefly or any iax phone. My problem is when i receive a call on my iax phone, the ring sound is very distort and bad. If i open my sip phone, and receive a call from my pstn, the ring is like dring dring, very normal. Otherwise, it is like a machine gun with iax Help would be really appreciate on how i can fix my iax issue JF Thank ___ 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 ___ALL-NEW Yahoo! Messenger - so many all-new ways to express yourself http://uk.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] FW: question about prepaid app_prepaid
Thanks to the lack of documentation, I decided to write my own AGI script (working but no where near complete) Look forward to replies and guidence on this topic. Umar. --- Yang Tao [EMAIL PROTECTED] wrote: Hi, I have compiled and installed app_prepaid module. But have problem when connect to postgres database. I guess so because after key in card number, it always play prepaid-no-aaa voice file. Anyone succeeded in configuring the app_prepaid for prepaid calling service for asterisk? Please help. Ps: where can I view the log file for this module. Thanks. Tom ___ALL-NEW Yahoo! Messenger - so many all-new ways to express yourself http://uk.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] Dyn Exten
What I do is create a file from setting in a DB. These extensions are included in the extensions.conf with a #include line. I can be done with a little perl scrit and a cron. After recreating the file all you have to do is reload the extensions. For eficiency, I create a temp file, and diff from the previous version (so I don't reload if I don't have to). On Thu, 2004-06-10 at 11:10, Jose R. Ortiz Ubarri wrote: Why??? Is there another way to do Dynamic Extensions??? -- JO Jeremy McNamara wrote: Jose R. Ortiz Ubarri wrote: Hi: Is DynExtebDB module still working?? Don't bother. That application should have never been written. Jeremy McNamara ___ 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 -- Pablo Endres [EMAIL PROTECTED] ComVoz Communications USA: +1 954 343-2085 Ext 199 Venezuela: +58 212 7713195 Ext 199 Colombia: +57 1 3256840 Ext 199 ___ 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 Apple PPC with YDL
Very interesting, Would like to hear what sort of performance you get out of it. I was considering linux on a sun box. Anyone done that ? Umar. --- Darren Sessions [EMAIL PROTECTED] wrote: Fyi, Successfully compiled Asterisk on an Apple G4 PPC with Yellow Dog Linux - without any source modifications. Worked fast and smooth. - Darren ___ 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 ___ALL-NEW Yahoo! Messenger - so many all-new ways to express yourself http://uk.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] Automating calls
Yes you can, I have never used it but here is a link http://www.voip-info.org/wiki-Asterisk+auto-dial+out Umar --- Simon [EMAIL PROTECTED] wrote: Hello I have heard that i can put a file in a certain directory to get * to initiate a call. Is this true ? if so where would i look ? Best Regards Simon Garvey ___ 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 ___ALL-NEW Yahoo! Messenger - so many all-new ways to express yourself http://uk.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] Mine strangest asterisk problem ever ....
Alessio Focardi wrote: BF You can try doing different things with it, but I know that I am currently BF set to level 3 rather than 5 as default with RedHat. I checked hdparm googling around, what parameter have you set to 3 instead of 5 ? I'm pretty sure this is a confusion. I think this must refer to runlevel 5 - 3 i.e. not having X running... F ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GSM to ISDN or TAPI
Storer, Darren wrote: Hi Chris, CL All I want is two GSM lines that look like voice modems to CL the PC and provide full telephony interface, that is DTMF CL both ways CLI and a few other bits and pieces. We use the Nokia 22: http://www.nokia.com/nokia/0,,56024,00.html They have worked well providing both telephony applications on remote sites and SMS support for Broadcast work in the UK (serial AT command interface). If you don't mind single band (900 or 1800 MHz GSM) operation there is an older device (Nokia Premicell) that can be sourced cheaply from eBay: http://www.nokia.com/cda1/0,1080,2700,00.html Does the incoming DTMF and voice work over the serial interface with the 22? I had a Nokia 32 for test and could not get it to return DTMF, it has AT commands to generate DTMF and to receive CLI but I could not get it into voice mode or get DTMF out of it. Thanks for your help Chris. ___ 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] Automating calls
Hi Simon, SG I have heard that i can put a file in a certain directory SG to get * to initiate a call. Is this true ? if so where SG would i look ? It *really* is time that you got to grips with voip-info.org. There are many gems in there; I typed in auto dial out and pressed the search button, have a look at what came back: http://www.voip-info.org/wiki-Asterisk+auto-dial+out ;-) HTH Darren -- Comgate TelcoInternetBroadcast -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Simon Sent: 10 June 2004 16:28 To: Asterisk-Users Subject: [Asterisk-Users] Automating calls Hello I have heard that i can put a file in a certain directory to get * to initiate a call. Is this true ? if so where would i look ? Best Regards Simon Garvey ___ 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] Automating calls
The directory is /var/spool/asterisk/outgoing. see http://www.voip-info.org/tiki-index.php?page=Asterisk%20auto-dial%20out for information on how to use the auto-dial out features... Steve Rosebush [EMAIL PROTECTED] PSTN: 1-248-724-4452 FWD: 63420 IAXTEL: 1-700-356-6191 -- all extension 201 on the PBX -- Simon wrote: Hello I have heard that i can put a file in a certain directory to get * to initiate a call. Is this true ? if so where would i look ? Best Regards Simon Garvey ___ 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] BT is moving to IP ONLY
Hi, all This is certainly very good news! http://www.neowin.net/comments.php?id=21119category=main ___ 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] Automating calls
Simon wrote: Hello I have heard that i can put a file in a certain directory to get * to initiate a call. Is this true ? if so where would i look ? The WIKI: http://www.voip-info.org bookmark it Jeremy McNamara ___ 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] Automating calls
/var/spool/asterisk/outgoing On Thu, 2004-06-10 at 15:27, Simon wrote: Hello I have heard that i can put a file in a certain directory to get * to initiate a call. Is this true ? if so where would i look ? Best Regards Simon Garvey ___ 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] Automating calls
On Thu, 2004-06-10 at 16:27 +0100, Simon wrote: Hello I have heard that i can put a file in a certain directory to get * to initiate a call. Is this true ? if so where would i look ? The rumours are true! You would look in the ever-so-helpful Wiki: http://www.voip-info.org/tiki-index.php?page=Asterisk%20auto-dial%20out HTH. Best regards, Gonzalo ___ 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] incoming DTMF on iConnectHere?
Hi, Anyone having problems receiving DTMF on incoming iConnectHere lines? They disappeared for us sometime in the last 12 hours... And, yes, we've restarted * and rebooted our * machine. Michael Swan Neon Software, Inc. ___ 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] Automating calls
Look in the asterisk source directory for a file called sample.call Read it and it'll give you all thed details -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Simon Sent: Thursday, June 10, 2004 10:28 AM To: Asterisk-Users Subject: [Asterisk-Users] Automating calls Hello I have heard that i can put a file in a certain directory to get * to initiate a call. Is this true ? if so where would i look ? Best Regards Simon Garvey ___ 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] Automating calls
On Thu, 2004-06-10 at 10:27, Simon wrote: Hello I have heard that i can put a file in a certain directory to get * to initiate a call. Is this true ? if so where would i look ? Google would have been a good starting point. Next would have been to exercise some curiosity in the source directory. Either way, you are looking for sample.call -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with * not detecting hangup on FXO and VM going into an infinite loop
On Thu, 2004-06-10 at 10:54, Chris Hirsch wrote: Hi everybody... I'm having an odd problem with voice mail on a recent CVS of * where it appears not to detect a hangup on FXO and * will keep treating the call as new and continue leaving voicemails until the max has been reached. It will then continue trying to leave voice mails and basically makes the system unavailble to any further incoming or outgoing calls on that FXO..has anybody seen this and if so how do I fix it? I've looked around on google and the list archives and it appears that there are others with similar problems with most people believing it to be a configuration problem. Since I don't see any bugs that have been formally posted with this description I think it most likely is...can anybody help me determine which option would be causing this behavior? I assume its in zapata.conf? If you where on google and saw the same questions, you should have been able to follow the rabbit further down the hole by playing with the links at the bottom of the pages google provided. You would have seen use rant regularly that you need to work just a tad harder to find the answer. It is there. It is consistently the same problem. You lack disconnect supervision. The only way to know the line has been hungup is to use progress detection and possibly tweak for your location. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Another Firefly update - now with SRV support
Adam Hart wrote: I've also added support for SIP via TCP and the ability to change the SIP port It complains every time you click OK in the Options page about Changing SIP port requires restart, even if you never looked at the SIP page (and don't even have any SIP networks configured). ___ 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: Automating calls
hi, Simon, you can look at this http://www.voip-info.org/tiki-index.php?page=Asterisk%20auto-dial%20out Simon wrote: Hello I have heard that i can put a file in a certain directory to get * to initiate a call. Is this true ? if so where would i look ? Best Regards Simon Garvey ___ 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] Dialing delay when using Zap channels
Good day, I've got around to installing an X100P card in my computer to try out asterisk. I noticed (and people who were testing with me also noticed) that when dialing from my SIP soft phone to the PSTN, the ringer tone changes after 2-3 seconds, precisely when the Zap channel takes over the call. Is it possible to eliminate the first ringing? Is there a reason to this delay-before-choosing-a-channel? Thanks, -- === Mathieu Nantel - RHCE,BOFH Ecopia BioSciences Systems Manager (514) 336-2724 x434 [EMAIL PROTECTED] === [*] Please avoid sending me Word/Excel/PowerPoint attachments: this assumes that I run MS Office, which is not always the case. See: http://www.fsf.org/philosophy/no-word-attachments.html === ___ 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] NAT and symmetric fw
Hi, I read to use NAT clients behind firewalls to use nat=yes and qualify=xxx to keep a nat connection open. I am using sip clients behind a firewall that is symmetric and my * box on public internet. Are these the only two options that I need in my configuration? Isnt the qualify command actually the amount of time it waits when it sends out the keepalive packet to the client before determining that its unreachable. How often does * send this keepalive packet? Is it configurable? Does anyone have this working with clients behind symmetric firewalls? If i turn off qualify, I am able to make calls to the sip client, but after 60 secs of inactivity calls made to the client time out, due to firewall closing nat connection. I dont understand why i can dial to the phone, but the keepalive fails right away. any ideas what would cause this? --- Harold Workman CCNA, CCNP Cytel Communications [EMAIL PROTECTED] Ph. 281-449-4000 x3098 ___ 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: Problem with * not detecting hangup on FXO and VM going into an infinite, loop
I think the problem it's in your dialplan, extensions.conf: ; voicemail management [voicemail] include = misc exten = 6245,1,VoiceMailMain2() exten = 6245,2,Hangup() Check the last line, I have the same problem and was because I wrote wrong the Hangup instruction... Regards! Date: Thu, 10 Jun 2004 09:54:32 -0600 From: Chris Hirsch [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Problem with * not detecting hangup on FXO and VM going into an infinite loop Reply-To: [EMAIL PROTECTED] Hi everybody... I'm having an odd problem with voice mail on a recent CVS of * where it appears not to detect a hangup on FXO and * will keep treating the call as new and continue leaving voicemails until the max has been reached. It will then continue trying to leave voice mails and basically makes the system unavailble to any further incoming or outgoing calls on that FXO..has anybody seen this and if so how do I fix it? I've looked around on google and the list archives and it appears that there are others with similar problems with most people believing it to be a configuration problem. Since I don't see any bugs that have been formally posted with this description I think it most likely is...can anybody help me determine which option would be causing this behavior? I assume its in zapata.conf? Thanks! Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GSM to ISDN or TAPI
I use Nokia 32s. I don't know what a fritz card is but they can act either as an FXO or as an FXS device. Beware though, if you use them off a port that expects a telephone set (like an ATA or so) you'll need a special cable to program the 32 properly - the cable is pricey.Acting asanFXO, they perform out of the box (don't need the cable). - Original Message - From: Chris Lee To: [EMAIL PROTECTED] Sent: Thursday, June 10, 2004 17:49 Subject: [Asterisk-Users] GSM to ISDN or TAPI HiI am in the UK and am looking for a device that will allow me to connect two sim cards (read wireless lines) to either the port on the back of my fritz card or any other connection direct to the PC that provides a usable telephony interface.I will even plug two devices into a windows box and have that do ISDN to ISDN if required.All I want is two GSM lines that look like voice modems to the PC and provide full telephony interface, that is DTMF both ways CLI and a few other bits and pieces.I am looking to using asterisk as a remote IVR for looking after some equipment, but land lines are a problem.Any help is much appreciatedRegardsChris.___Asterisk-Users mailing list[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] mysql errors
Since updating * via CVS earlier this week, I've been having problems with cdr_mysql. Prior to that time my queries and cdr all worked fine. Now, even though my queries still work, I get the messages similar to this: ERROR[1211374384]: cdr_addon_mysql.c:203 mysql_log: Failed to insert into databas-- Hungup 'Zap/74-1' Anyone have any suggestions what went wrong? ___ 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 as a VoIP Gateway to an Analog PBX
Hello all, I am a relative asterisk noob so please bear with me if my questions are obvious. What I'm trying to do is get our analog PBX (A Merlin Legend) connected to VoIP. From all my googling and reading voip-info.org (and this list) it seems very possible. I just wanted to describe my setup and see if I'm going in the right direction. What I'd like to do is set up an asterisk box with a T405P Quad-Span T1 Card. I am planning to drop all voice lines and switch to a full data T1. Span 1 would be pure data (PPP encapsulation) coming from the Telco. Span 2 would be all voice channels, going into an CAC Adit 600 Channel Bankwith 3 FXS cards and from there into the PBX. I would really like to use a provider who supports IAX2 termination like NuFone, or VoicePulse but VoicePulse dosen't have a local DID (We're located in Portland, Oregon) and it dosen't look like they provide (8XX) numberservice. NuFone has a nice website but absolutely NO info on their rates/services/etc... when I call I just get music so I don't entirely trust that they'll always be there. If anyone can recommend a good IAX2 service that would be excellent! I'm thinking about just using iconnecthere.com with a SIP connection for now... From what I understand, I can set up all the FXS (Asterisk FXO - Adit 600 FXS) channels into a groupin zapata.conf and when I reference that group with a dial command (like Dial(Zap/g1,seconds,options), Asterisk will "Hunt" that group for a channel that is not busy? When a call comes into Asteriskvia the DID (through HDLC(n) interface) it will then ring into the extension I register it to in sip.conf (e.g. register = myusername:[EMAIL PROTECTED]/1234). Then I can create a definition [host.sipprovider.com] and set it's context to something like [sip-in] through extensions.conf. Then in the [sip-in] context, I can tell it to ring into the channel group in span 2 that I've created and it will automatically hunt for a free FXS channel? I could even set it upwith MusicOnHold in case all the PBX channels are busy... Am I right in this? Will this work? Hopefully I have given enough information, please let me know if I need to explain something further. I really appreciate any input you have on my plan. P.S. Asterisk is AWESOME, It's been a long time since I've been this excited about a new application coming out. I believe technology like this will revolutionize the internet in short order. -Thanks in advance Chris Chris ShawIS ManagerWater Tech IndustriesPhone: (888)-254-8412Fax: (503)-261-9118E-Mail: [EMAIL PROTECTED]
Re: [Asterisk-Users] Dyn Exten
Pablo Endres wrote: For eficiency, I create a temp file, and diff from the previous version (so I don't reload if I don't have to). Why bother doing that much processing? Just set a flag somewhere that determines weather or not you need a reload. Jeremy McNamara ___ 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] BUG?: reinvite and nat
I have a setup where I am using asterisk a SIP proxy. My ATA is behind NAT. Asterisk user is setup with nat=1 and canreinvite=yes. The call sets up and I can get one way media. The media works ok from the ATA behind the NAT to the external SIP endpoint, but I cannot get media back through the NAT. Calls work just fine if I call from the ATA into an asterisk IVR menu, so I know the NAT router is working properly. Also, if I set canreinvite=no, then things work ok. I see some strange ReINVITEs happening that seem to be the cause of this. When the ReINVITE is sent to the external SIP proxy, it puts SDP parameters out of the raw SDP supplied by the ATA (internal IP), but then issues a second ReINVITE with the correct external NAT IP (detected from the packet source address), but a few moments later it send a 3rd ReINVITE back to the internal NAT IP. What I think is happening is when asterisk sent the ReINVITE toward the ATA, when the ATA issued the 200 response with its SDP, by that time asterisk had sent the 2nd ReINVITE (external IP) to the external SIP endpoint, the 200 reply from the ATA had the internal NAT IP, which was different than it just transmitted to the remote endpoint, so it thought it had to send another 3rd ReINVITE, but this had the internal NAT IP, so it broke media into the NAT from the remote UA. So, I think the bug is that asterisk is sending this 3rd ReINVITE when it should not. I have a trace of all the SIP messages here: http://www.cheapnet.net/~mike/asterisk_excel_with_reinvite.log This is a complicated issue, hopefully I explained it well. In that SIP trace file, the remote SIP UA is 172.20.50.30 (media to .32 and .33), asterisk is 172.20.50.22. The NAT box is 10.10.11.77 on the external interface and 192.168.222.1 on the internal NAT side. The ATA is 192.168.222.197. Between 10.x and 172.x is straight routing (this is internal test network). The only nat is 192.168.222.x is translated to 10.10.11.77 to reach any of the 172.x network. ___ 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] BT is moving to IP ONLY
See also: http://www.adslguide.org.uk/newsarchive.asp?item=1723 - Original Message - From: Senad Jordanovic [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, June 10, 2004 5:18 PM Subject: [Asterisk-Users] BT is moving to IP ONLY Hi, all This is certainly very good news! http://www.neowin.net/comments.php?id=21119category=main ___ 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] BT is moving to IP ONLY
Their Syntegra trading turrets already have begun the migration. Now, if I can get my hands on one and getting it to work with *, I'll be set. -cj -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Senad Jordanovic Sent: Thursday, June 10, 2004 12:18 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] BT is moving to IP ONLY Hi, all This is certainly very good news! http://www.neowin.net/comments.php?id=21119category=main ___ 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] New version of DIAX (0.9.8a) available nowfor free download
On Thu, 2004-06-10 at 02:04, Dan wrote: Unfortunately not, because the GSM phone does not support Audio Gateway profile (just Headset profile). It can connect only with the headset. ..but.. you can use the Bluetooth headset for DIAX and the GSM phone as CallerID/Dialer. .. and all this even when the computer screen is locked. I tried the Bluetooth stuff yesterday and I was able to input digit but I didnt found the Enter o Dial key Im using a SE t610. -- Juanjo sin .sig ___ 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] Mine strangest asterisk problem ever ....
BF You can try doing different things with it, but I know that I am
currently
BF set to level 3 rather than 5 as default with RedHat.
I checked hdparm googling around, what parameter have you set to 3
instead of 5 ?
Alessio Focardi wrote:
I'm pretty sure this is a confusion.
I think this must refer to runlevel 5 - 3
i.e. not having X running...
F
No, no confusion. If I do a hdparm -i /dev/hda I receive:
/dev/hda:
Model=Maxtor 6E040L0, FwRev=NAR61590, SerialNo=E1JN7RXE
Config={ Fixed }
RawCHS=16383/16/63, TrkSize=0, SectSize=0, ECCbytes=57
BuffType=DualPortCache, BuffSize=2048kB, MaxMultSect=16, MultSect=16
CurCHS=16383/16/63, CurSects=16514064, LBA=yes, LBAsects=78165360
IORDY=on/off, tPIO={min:120,w/IORDY:120}, tDMA={min:120,rec:120}
PIO modes: pio0 pio1 pio2 pio3 pio4
DMA modes: mdma0 mdma1 mdma2
UDMA modes: udma0 udma1 udma2 *udma3 udma4 udma5 udma6
AdvancedPM=yes: disabled (255) WriteCache=enabled
Drive conforms to: (null): 1 2 3 4 5 6 7
The UDMA mode was changed (lowered) from 5 to 3. I am not sure of the
syntax to do it, however man hdparm will lead you in the right way.
- Brent
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Re: [Asterisk-Users] Presence
This is a wonderful idea. I like the app_im concept a lot. I'd make a few additions though. Like the ability to have festival read the Away message as the Voicemail message. I'd definitely change my voicemail more often if I could do it by changing my Jabber away message. I would suggest that Jabber would be a more effective first target though, as with it comes the ability to hit AIM/ICQ/MSN/Yahoo/etc users via a simple proxy. Having just the one implementation would simplify things. Chris Tooley On Wed, 2004-04-07 at 21:29 -0400, John Todd wrote: At 8:29 PM -0400 on 4/7/04, Shad Mortazavi wrote: I have to agree. A large number of people are looking for this feature. I have written a web script that can show Agent logged into the system. I think integration/gateway between Asterisk and Jabber would be a amazingly wonderful product. There is always MSN. Shad Mortazavi --- Nexus Technical Manager n|m Nexus Management Inc Netural Bay Sydney My idea for AIM/Jabber/Yahoo integration is below. Comments and/or programmers are welcome to have at it, and to expand on my ideas. I have mentioned this to several programmers who expressed an interest, but I'm sure that lack of time and funding has kept them from starting on the project, if it indeed is worthwhile. This is a kludge to some degree, but it uses _already existing_ presence tools to extend Asterisk's functionality, without needing to modify any client software or hardware. This is really a one-way presence idea at the moment. There are the glimmerings of two-way presence (see the activewhen keyword) but this is mostly for CTI outbound notices from an * server to humans upon some events defined by the administrator. I would see this most typically used either as a screenpop on an inbound or outbound call, or perhaps as a voicemail notification tool if the administrator is clever enough to embed a URL into the string for the instant message text. Phase 1: Create a set of programs for Asterisk which allows status checking of a particular username on a particular instant messaging system (availability, idle time) and also allows for transmission of instant messages from Asterisk to other users on those instant messaging systems (one-way.) The first systems that come to mind would be AOL's AIM and Yahoo. Phase 2: Add additional instant message systems: maybe Jabber, MSN. Allow examination of user's header line (in AOL, at least) and pass that through the app_imstatus return codes. This would allow me to specify mobile: as the first digits of my status, thus a GotoIf would be able to know that it should send calls to my cell phone. Or when I get to work, and shift between my home account (home: hello, I'm home) to work (work: at my desk) then the system will automatically forward calls appropriately. This might be easy enough to do in Phase 1, but I'm uncertain. Future paths: A true presence application for telephony in a large scale method is lacking today. It may be the case that this could be done by creating a custom telephony presence presentation application that is based on an existing (or multiple existing) chat protocols. As an example, it is possible that I might be able to make my status message on AIM change from avail/sip:[EMAIL PROTECTED] to busy/sip:[EMAIL PROTECTED] every time I pick up the phone; that could be done programmatically by Asterisk. Then, my friends who have the custom telephony presence application would see the little icon beside pinkycaruthers go from green to flashing orange. As soon as I went back to non-busy, they could just click on my icon, and two things would happen: a password-protected message would get fired off to THEIR phone system and extension from the presence application on their desktop, which in turn would be received by an asterisk-aware application on their Asterisk server, which in turn would create a spool call to MY phone system from the SIP URI that I included in my Status message. Presto! We have minimalist call routing, presence, and click-to-dial - we're just missing the little app to do it on Windows, MacOS, Linux, Java, whatever. The core message transport protocols all exist; it's just a matter of layers on top of them. Using standard telephony URI's, we could not just do this with SIP, but with tel, h323, iax2, anything - it's not limited to VoIP. ; im.conf ; ; Use of this file implies that you have an active account with one or more ; instant messaging services, and that you probably use an account that is ; dedicated to your Asterisk server so it knows what's going on. You may ; need to ensure that any other user id's that you expect to receive messages ; are filtered in such a way that the messages from your Asterisk-specific ; account are permitted
Re: Re: [Asterisk-Users] Iax2 ringtone problem
yes the bell to notify, when it is to iax, the bell sound is very bad. With sip it's fine, the ringback is good with both technology Regards JF De: Umar Sear [EMAIL PROTECTED] Date: 2004/06/10 jeu. PM 12:00:33 GMT-04:00 À: [EMAIL PROTECTED] Objet: Re: [Asterisk-Users] Iax2 ringtone problem Hi Jean, It seems that no one on the list is interested in IAX, I have posted a couple of basic questions but no ones seems to want to answer. I guess everyone is busy right now. Anyway back to your question. When you say the ringtone , do you mean the rinback tone (what the caller hears) or bell to notify the callee that there is a call. Umar. --- Jean-Francois Dubé [EMAIL PROTECTED] wrote: Hi, i have a problem with iax2 and ringtone. Here is the call path pstn - asterisk - iax - firefly or any iax phone. My problem is when i receive a call on my iax phone, the ring sound is very distort and bad. If i open my sip phone, and receive a call from my pstn, the ring is like dring dring, very normal. Otherwise, it is like a machine gun with iax Help would be really appreciate on how i can fix my iax issue JF Thank ___ 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 ___ALL-NEW Yahoo! Messenger - so many all-new ways to express yourself http://uk.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 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 Apple PPC with YDL
On Thu, 2004-06-10 at 12:04, Umar Sear wrote: Very interesting, Would like to hear what sort of performance you get out of it. I was considering linux on a sun box. Anyone done that ? It surely gives a new life to those old suns, usually linux runs faster than slowlaris. I have installed on a Sparcstation 4/5, Ultra 1, Ultra 10 and E450 even managed to get a Javastation to boot. -- Juanjo sin .sig ___ 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] incoming DTMF on iConnectHere?
At 09:43 AM 6/10/2004 -0700, you wrote: Hi, Anyone having problems receiving DTMF on incoming iConnectHere lines? They disappeared for us sometime in the last 12 hours... And, yes, we've restarted * and rebooted our * machine. Hi, I'll follow up on my own question. :-) Here is the response from iConnectHere customer support: Please be advised that we do not support PBX system for receiving DTMF sounds. We understand that it was working for you in the past, but it is not a function that our network is set up to support. A follow on email from iConnectHere implies the recent changes to their voicemail system have broken DTMF on incoming calls to *. They offered me $1 credit as the result of my inconvenience. :-( Michael Swan Neon Software, Inc. ___ 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
