Re: [asterisk-users] Digium through Octasic
Is there a trade-in program in place ? I have a te410p and a te405p that I am not using because of various problems we had, but would like to try the te407 ... Julian BJ Weschke wrote: On 11/30/06, Andrew Kohlsmith [EMAIL PROTECTED] wrote: On Thursday 23 November 2006 11:44, Heidi Mendoza wrote: We're looking at using 4 or 8 port T1 cards with echo cancellation and are evaluating brands to go with. We know that Sangoma has excellent solutions especially when it comes to echo. But we still have to hear about actual performance of a Digium card using the same Octasic DSP echo canceller. Excellent performance. I had an A104d which was giving some very odd audio artifacting, Sangoma replaced the card but did not test the original to ensure that the card was indeed defective or that the problem was solved with the replacement. I haven't put the replacement in service yet, as I had a TE407P on order and it arrived first. :-) After dealing with the crap that the TE406P was, the TE407P is *heaven*. Highly recommended. Ditto here as well. The TE412P and TE212P have been rock solid in deployments I've put them in to. Kudos to the Digium folks for getting it right here. They've got a great product that I wouldn't hesitate to recommend with this product line. BJ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zaptel compilation problems with linux 2.6.19
Hi On Sat, Dec 02, 2006 at 07:17:22PM -0500, Matthew Rubenstein wrote: On Sat, 2006-12-02 at 09:53 -0700, [EMAIL PROTECTED] wrote: Date: Sat, 2 Dec 2006 11:51:37 +0200 From: Tzafrir Cohen [EMAIL PROTECTED] Subject: Re: [asterisk-users] zaptel compilation problems with linux 2.6.19 To: Asterisk-Users asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii Hi Hi, and thanks for the help :). On Fri, Dec 01, 2006 at 01:43:20AM -0500, Matthew Rubenstein wrote: On Thu, 2006-11-30 at 17:56 -0700, [EMAIL PROTECTED] wrote: Message: 18 Date: Fri, 1 Dec 2006 00:56:10 +0200 From: Tzafrir Cohen [EMAIL PROTECTED] Subject: Re: [asterisk-users] zaptel compilation problems with linux 2.6.19 To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii On Thu, Nov 30, 2006 at 02:44:03PM -0500, Matthew Rubenstein wrote: I'm having problems installing ztdummy on my CentOS-2.6.16-rc6/Asterisk-1.2.11 datacenter server (no ZAP HW, SIP only to PSTN). I unpacked the kernel sources and headers in a directory, made (but not re/installed) the kernel, unpacked the zaptel-1.2.11 tarball, then went thru the make sequence. It seemed to proceed OK (without errors, just some warnings), but didn't seem to result in a loadable ztdummy kernel module. Complete (failed) install session transcript is attached to this message; details appended: - # cd path-to-zaptel-1.2.11-source # export KSRC=path-to-kernel-source-root-dir # make clean # make config [... series of shell script conditionals apparently executed OK ...] # make linux26 [... series of CC/LD reports, some warnings, no errors ...] # make install [... series of INSTALL messages, same warnings from (make linux26), no errors ...] # modprobe ztdummy FATAL: Module ztdummy not found. FATAL: Error running install command for ztdummy # modprobe zaptel FATAL: Module zaptel not found. - (make linux26) generated some warnings about various usb_*_dev symbols undefined in [xpp,wcusb]/*.ko, but no actual errors. (make install) Those are harmless, IIRC. I'll try to recall their source. I suspected as such. But I don't think the server has full USB/UHCI support running, or fully installed: - # lsmod Module Size Used by binfmt_misc12168 1 dm_mod 59512 0 thermal13864 0 processor 25284 1 thermal fan 4772 0 floppy 63172 0 generic 4836 0 [permanent] ide_generic 1504 0 [permanent] # modprobe usb_uhci FATAL: Module uhci_hcd not found. # modprobe uhci FATAL: Module uhci_hcd not found. - repeated those warnings. (modprobe ztdummy) finished with Was depmod run? No, but trying it now (after the transcripted session) didn't seem to help: - # depmod # modprobe ztdummy FATAL: Module ztdummy not found. FATAL: Error running install command for ztdummy - uname -r # uname -r 2.6.16-rc6-060427a so depmod, modprobe and such will look under /lib/modules/2.6.16-rc6-060427a , but the modules were installed elsewhere: [ snip: they are under /lib/modules/2.6.16-rc6/misc ] One way to fix this is to move the modules, or pass the kernel vesiosion explicitly to make with KVERS . However this raises the question: does the kernel source tree you used to build the module matches the running kernel version. Is it a kernel you have built? Is there a link /lib/modules/2.6.16-060427a/build ? The build link is normally generated automatically by the install target of the kernel. Are you sure it is a good idea to mess with it? Why not just build zaptel with the kernel tree that was used to build the kernel? Where is that kernel from? Have you built it?Doesn't look like a CentOS kernel. OK, I did the following: # cd /lib/modules/2.6.16-rc6-060427a # ls -l build source lrwxrwxrwx 1 root root26 Sep 10 08:54 build.orig - /home/src/linux-2.6.16-rc6 lrwxrwxrwx 1 root root26 Sep 10 08:54 source.orig - /home/src/linux-2.6.16-rc6 # mv build
Re: [asterisk-users] Detailed description of problem in Poland
On 2 Dec 2006, at 16:59, Alex Rixhardson wrote: Hi guys, Here is a bit more detailed information of my problem: If I connect Asterisk PBX to the Polish telco via E1, I don't get any red alarms or anything. The line seems to be fine and the inbound calls are also accepted by the Asterisk. However, whenever I try to make an outbound call, the call is either stuck (Asterisk just displays Called g1/482 and then nothing), or I get the following message: -- Called g2/ -- Channel 0/1, span 3 got hangup -- Zap/63-1 is circuit-busy -- Hungup 'Zap/63-1' == Everyone is busy/congested at this time (1:0/1/0) The telco guys say that my request to make an outbound call is missing a RING message. What must be set in the zapata.conf or zaptel.conf to make Asterisk send RING message? Does anyone have any sample zapata.conf or zaptel.conf for connection between Asterisk to Polish telco via E1? I don't think (based on the debug output in your other email) that it is getting to the 'ring' phase. the telco rejects the call as having a protocol error in the setup message. They don't like one of the info elements in your setup message. You could try playing with the TON and calling name fields. ( pridialplan= usecallerid= hidecallerid= usecallingpres= in zapata.conf ) Tim. Any suggestions will be most appreciated, Alex Check out the all-new Yahoo! Mail beta - Fire up a more powerful email and get things done faster. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Tim Panton www.mexuar.net www.westhawk.co.uk/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RNK
Hi List, I just wanted to let eveyone that RNK uses asterisk. One more big company to let clients know that asterisk is great.___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] translate.c:88 powerof: Powerof 0: No power?? / translate.c:133 ast_translator_build_path: No translator
Dec 2 17:45:19 WARNING[64722]: translate.c:88 powerof: Powerof 0: No power?? Dec 2 17:45:19 WARNING[64722]: translate.c:88 powerof: Powerof 0: No power?? Dec 2 17:45:19 WARNING[64722]: translate.c:133 ast_translator_build_path: No translator path from gsm to unknown Dec 2 17:45:19 WARNING[64722]: translate.c:88 powerof: Powerof 0: No power?? Dec 2 17:45:19 WARNING[64722]: translate.c:88 powerof: Powerof 0: No power?? Dec 2 17:45:19 WARNING[64722]: translate.c:133 ast_translator_build_path: No translator path from gsm to unknown I have been getting these mostly on zap calls, but it has been showing up on sip calls too. Anyone have any ideas on this? According to all the bug reports, this was supposed to be fixed in 1.2.13 Asterisk 1.2.13 FreeBSD 5.5-RELEASE-p4 asterisk-1.2.13 An Open Source PBX and telephony toolkit asterisk-addons-1.2.3_1 Additional modules for the Asterisk Open Source PBX zaptel-1.0_1A FreeBSD Driver for FXO, FXS, BRI and PRI Telephony Cards libpri-1.2.3A C implementation of the Primary Rate ISDN specification signature.asc Description: OpenPGP digital signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RTP Media Path
I know this has been asked before and I went over the wiki but I have not been able to come to a clear answer. 1) If I have SIP Provider Asterisk - ATA and vice versa (ATA - Asterisk SIP Provider) from what I understand if NO NAT is being used then asterisk just starts and stops the session however the RTP media stream will be passed directly from the SIP provider and vice versa. (This is of course if there is no NAT involved). Now say I had such a set up will the server be able to handle more calls than average if the only responsibility if the server is to authenticated and pass along the calls ? (There will be an AGI running in the begining to determine what route to used based on how many minutes each route has used). Now if the ATA's are behind VOIP and asterisk is on a public IP then does asterisk have to sit in the media path ? Also can some one explain exaclty when the RTP session is started and stopped. Also another set up we are woroking on is SIP Provider (Incoming DID) Asterisk (for authentication based on PIN) - Back to SIP Provider. The asterisk server will be on a public IP. Can I have asterisk stay out fo the media path (here I asume yes. Just wana be 100% sure). Thanks a lot. Dovid___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] G729 Liscence
If I have SIP Provider (sending call as G729) Asterisk (picks up and asks for PIN) --- call sent to SIP provider I understand that I will need a g729 liscence for when the user needs to enter thier PIN number. Once the call is handed off to the SIP provider will I still need a liscence for the call or will RTP session go between the providers and I will be able to free up a G729 liscence ? Thanks. Dovid ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RTP Media Path
Asterisk wont sit in media path if both callee and caller agrees on common codec, both have canreinvite=yes in sip.conf, no t,T are used in dialplan ( please correct me if i am wrong ) , no call recording is enabled . I think asterisk does native bridging even if one is behind nat ( i tested with atleast one party behind nat not sure if it works when both are behind nat ) and devices should support reinvites .. On 03/12/06, Dovid B [EMAIL PROTECTED] wrote: I know this has been asked before and I went over the wiki but I have not been able to come to a clear answer. 1) If I have SIP Provider Asterisk - ATA and vice versa (ATA - Asterisk SIP Provider) from what I understand if NO NAT is being used then asterisk just starts and stops the session however the RTP media stream will be passed directly from the SIP provider and vice versa. (This is of course if there is no NAT involved). Now say I had such a set up will the server be able to handle more calls than average if the only responsibility if the server is to authenticated and pass along the calls ? (There will be an AGI running in the begining to determine what route to used based on how many minutes each route has used). Now if the ATA's are behind VOIP and asterisk is on a public IP then does asterisk have to sit in the media path ? Also can some one explain exaclty when the RTP session is started and stopped. Also another set up we are woroking on is SIP Provider (Incoming DID) Asterisk (for authentication based on PIN) - Back to SIP Provider. The asterisk server will be on a public IP. Can I have asterisk stay out fo the media path (here I asume yes. Just wana be 100% sure). Thanks a lot. Dovid ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RTP Media Path
This is correct, if no NAT is involved anywhere and reinvites are allowed then Asterisk will stay out of the media path and be used only as Signaling server. So as for your answer yes, it will be able to handle more calls than expected because there is no CPU overhead of the media path. It is common strategy to have a single signaling server and have RTP servers all around the globe for latency and etc, media gateways. Vicky wrote: Asterisk wont sit in media path if both callee and caller agrees on common codec, both have canreinvite=yes in sip.conf, no t,T are used in dialplan ( please correct me if i am wrong ) , no call recording is enabled . I think asterisk does native bridging even if one is behind nat ( i tested with atleast one party behind nat not sure if it works when both are behind nat ) and devices should support reinvites .. On 03/12/06, *Dovid B* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I know this has been asked before and I went over the wiki but I have not been able to come to a clear answer. 1) If I have SIP Provider Asterisk - ATA and vice versa (ATA - Asterisk SIP Provider) from what I understand if NO NAT is being used then asterisk just starts and stops the session however the RTP media stream will be passed directly from the SIP provider and vice versa. (This is of course if there is no NAT involved). Now say I had such a set up will the server be able to handle more calls than average if the only responsibility if the server is to authenticated and pass along the calls ? (There will be an AGI running in the begining to determine what route to used based on how many minutes each route has used). Now if the ATA's are behind VOIP and asterisk is on a public IP then does asterisk have to sit in the media path ? Also can some one explain exaclty when the RTP session is started and stopped. Also another set up we are woroking on is SIP Provider (Incoming DID) Asterisk (for authentication based on PIN) - Back to SIP Provider. The asterisk server will be on a public IP. Can I have asterisk stay out fo the media path (here I asume yes. Just wana be 100% sure). Thanks a lot. Dovid ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] * key on Linksys SPA-841
I wonder if anyone has experienced an issue I have found with the Linksys SPA-841 phone. On my Asterisk (Trixbox 2), to login to a queue, a user must enter the queue number, followed by the * key. This works fine on my Companies mix of phones, with the exception of the Linksys (Sipura) SPA-841. David Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodatasolutions.com http://www.autodatasolutions.com/ Tel: (519) 963-3020 Fax: (519) 451-6615 Lead, follow or get out of the way! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Realtime fullcontact field contains nat device private ip
Hi All, Has anyone else noticed that when a sip phone sitting behind a nat registers to asterisk using realtime database, the private IP of the phone is put into the fullcontact field instead of the public contact IP. The database has the correct public IP in the ipaddr field and correct port number in the port field, which is actually what asterisk uses to to contact the device. This eliminates the ability to use the fullcontact URI to directly contact the nat'ed phone. Works great for non-nat'ed devices. Is this by purpose or an oversight the way Realtime pulls the correct contact info in the sip registration header from the device? Does anyone know how to correct this behavior? It is the same with nat=yes or nat=no. Thanks. JR -- JR Richardson Engineering for the Masses ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoIP GSM Gateways
Not very good at answering followups to your ads, are you, Sam? On 01/12/06, Peter Bowyer [EMAIL PROTECTED] wrote: On 30/11/06, Sam Tam [EMAIL PROTECTED] wrote: We do have @cough VoIP GSM Gateway for sell as well @ cough Try to search on ebay for gsm voip gateway and you will see some in there As far as I am concern it is cheaper than 2n. And if you are looking for multi ports then it will come off as RJ11 ports rather than voip and they are £100 per port with a max of 16 ports in 1 chassis. It's cheaper because it's not the same thing and only does half the job - what you sell is an analogue-GSM adapter. It needs an FXS port to interface with Asterisk, and isn't actually a VoIP GSM gateway at all. If you must plug it here, please be honest about what it is and what it's not. Peter -- Peter Bowyer Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] VoIP GSM Gateways
Have you looked at his website, www.netenable.co.uk ? Looks like he pays bills the same way as he answers followups ;-) g -Original Message- From: [EMAIL PROTECTED] on behalf of Peter Bowyer Sent: Sun 03-Dec-06 8:43 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] VoIP GSM Gateways Not very good at answering followups to your ads, are you, Sam? On 01/12/06, Peter Bowyer [EMAIL PROTECTED] wrote: On 30/11/06, Sam Tam [EMAIL PROTECTED] wrote: We do have @cough VoIP GSM Gateway for sell as well @ cough Try to search on ebay for gsm voip gateway and you will see some in there As far as I am concern it is cheaper than 2n. And if you are looking for multi ports then it will come off as RJ11 ports rather than voip and they are £100 per port with a max of 16 ports in 1 chassis. It's cheaper because it's not the same thing and only does half the job - what you sell is an analogue-GSM adapter. It needs an FXS port to interface with Asterisk, and isn't actually a VoIP GSM gateway at all. If you must plug it here, please be honest about what it is and what it's not. Peter -- Peter Bowyer Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This email has been scanned for all viruses by the Star Internet Virus Screen. The service is provided in partnership with MessageLabs, the email security company. For more information on a higher level of virus protection visit www.star.net.uk __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CAPI module issue
Hi List, I am experiencing an issue with a server running asterisk; I installed an AVM FRITZ card and configured it to work with the capi module. Once everything is installed the card works perfect; the issue is that every time I reboot the machine I have to re install the capi4k-utils before I can load asterisk otherwise the capi module does not loadup. After boot up when I try capiinfo I get the following error message, capi not installed - No such file or directory (2) Once re-installed I get the following, [EMAIL PROTECTED] ~]# capiinfo Number of Controllers : 1 Controller 1: Manufacturer: AVM GmbH CAPI Version: 2.0 Manufacturer Version: 3.11-02 (49.18) Serial Number: 101 BChannels: 2 Furhter details follow: Problem: capi4k-utils (capilib_new_ncci) does not load automatically after boot up AVM FRTIZ card latest driver (BRI) capi module: chan_capi-cm-0.6.5 capi4k-utils Linux Distro: Fedora core 3 Kernel: 2.6.12-2.3.legacy_FC3 (capi supported by kernel) Has any one experienced this before? Can anyone please provide with some ideas to overcome this issue? Thanks in advance Paul _ Advertisement: Meet Sexy Singles Today @ Lavalife - Click here http://a.ninemsn.com.au/b.aspx?URL=http%3A%2F%2Flavalife9%2Eninemsn%2Ecom%2Eau%2Fclickthru%2Fclickthru%2Eact%3Fid%3Dninemsn%26context%3Dan99%26locale%3Den%5FAU%26a%3D23769_t=754951090_r=endtext_lavalife_dec_meet_m=EXT ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.4: SPANDSP3 (WIP) HOWTO
Hi All, To coincide with my Asterisk 1.4 Beta Howto, I've also started the spandsp3 howto. The howto is functioning, however when attempting to use rxfax or txfax it crashes asterisk. I am not a programmer thus couldn't figure out how to make it not seg. Check out the WIP howto for more information, and diffed files. Anyone who knows what they are doing is welcome to help fix the howto, and the app_rxfax/app_txfax files. Comments welcome! http://www.voipphreak.ca/archives/357-Asterisk-1.4-Beta-SpanDSP3-Howto-WIP.html#extended or http://tinyurl.com/tnal5 Thanks, Diwelf ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk : Numbers Guessing Game
A while ago I wrote a numbers guessing game to keep me entertained on those really boring days :). I've uploaded it to the blog for the rest of you to enjoy or modify as well. It's a simple game to guess what number the PBX is thinking of and return a yay you got it right or a sorry that's wrong. All sound files are included and ready to add to your system in a few minutes. Check it out here: http://www.voipphreak.ca/archives/358-Asterisk-Howto-Numbers-Guessing-Game.html or http://tinyurl.com/sr4p4 Thanks, Diwelf ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] upgrading grandstream GXP-2000 from 1.0.2.13 to1.1.1.14
Thanks for your help Claudemir, I look forward to the response. Seems odd that they don't post an archive of their old firmware versions on their website, or at least ones that are required to get to the latest release from whatever is in the field already. Regards, Scott From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Claudemir F. Martins Sent: Saturday, December 02, 2006 11:16 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] upgrading grandstream GXP-2000 from 1.0.2.13 to1.1.1.14 Hi Scott, I have direct contact with a support person from Grandstream. I will ask him about that and tell you what did he say as soon as possible. Please just wait. Regards Claudemir On 11/30/06, Scott Keagy [EMAIL PROTECTED] wrote: So I've got phones with ancient firmware, and the release notes for 1.1.1.14 say read the previous release notes and first upgrade to 1.1.0.16 The 1.1.0.16 firmware is not available for download from the grandstream website (at least I haven't found it). Any pointers on where to get this intermediate image? I already tried googling to no avail (didn't help that I was using a link with 2000 ms latency). Plus, any overall pointers for making this upgrade process a success would be appreciated. Regards, Scott ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk manager originate command
Hi everybody, I want to know how to get the uniqueid or a call started from asterisk manager using Originate command. Best regards Rodrigo Gonzalez ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] G729 Passthru?
I have a SIP carrier which accepts only G729 connections from my Asterisk server. If all the server does is Dial() (out) two legs of a call which are natively bridged, with no processing the media (and no DTMF detection, etc), do I need to install a G729 codec of my own? All the media from each leg connected to the other is already encoded into G729 by the SIP carrier from which it's coming for feeding back to the SIP carrier. Does that loopback work without a G729 codec on the server? If not, what would the codec actually do with the data it gets? A related issue is whether I can pre-encode recorded audio files with a G729 codec. So the server can send wakeup call messages to the SIP carrier without running the codec at call time, just sending the pre-encoded media to the SIP carrier. -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Hold calling channel and ask called channelbeforeconnect???
Thanks. That looks exactly what I need. Will test in the next couple of days -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C F Sent: Saturday, December 02, 2006 8:54 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Hold calling channel and ask called channelbeforeconnect??? you can find an example on the wiki here: http://www.voip-info.org/wiki/view/Asterisk+cmd+dial On 12/1/06, Nigel J. Terry [EMAIL PROTECTED] wrote: I posted this a week ago and have had no response. Can someone tell me if I am asking a stupid question, i.e. is the answer either obvious or impossible? Thanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nigel J. Terry Sent: Wednesday, November 22, 2006 10:27 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Hold calling channel and ask called channel beforeconnect??? I am a newbie. Just got my Asterisk working and I love it. I want to do the following, believe it should be possible, but can't work out how: When I get an incoming call, I want to answer and just send ringing to the calling channel. Then I want to call the destination channel, send a message asking if they will accept the call, get a response (1 or 2) and then either connect the parties (1) or send the calling channel to voicemail (2). Any ideas, thanks Nigel ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Neat Skype Device
Not sure if anyone has posted this before, but it would be great (providing it works well) to connect this to an FXO port. http://www.jr.com/JRProductPage.process?Product_Code=AEE+USB07051C01JRSource=PriceGrabber.datafeed.AEE+USB07051C01 Anyone tried this product yet? Great price for a cool toy. Thanks, Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] TDM01B installation
Hi, iam new in this milis I have problem with TDM01B Installation, output zttool command is Unable to open /dev/zap/ctl: No such device or address and then i find the same IRQ uses VGA compatible controller and Communication controller is 169 What can i do next ? please your advice - This email was sent using Student EEPIS-Webmail. http://student.eepis-its.edu/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G729 Passthru?
I think you do need to buy the G729 for each call. If your system is using anything other than G729. That is the way I was told it works. But I don't use G729. Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email (VOIP PBX) 1-866-638-1254 (Voip PBX) Free World DialUp: 780-217 WebSite: http://www.freeworlddialup.com/ We have Toll Free DID's instock * * * NO MONTHLY FEE - LIMITED TIME ONLY * * * http://www.bochterservices.com/?t=TF(NM)did BUY Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email For new and used security items http://www.bochterservices.com/?j=storet=email_security Matthew Rubenstein wrote: I have a SIP carrier which accepts only G729 connections from my Asterisk server. If all the server does is Dial() (out) two legs of a call which are natively bridged, with no processing the media (and no DTMF detection, etc), do I need to install a G729 codec of my own? All the media from each leg connected to the other is already encoded into G729 by the SIP carrier from which it's coming for feeding back to the SIP carrier. Does that loopback work without a G729 codec on the server? If not, what would the codec actually do with the data it gets? A related issue is whether I can pre-encode recorded audio files with a G729 codec. So the server can send wakeup call messages to the SIP carrier without running the codec at call time, just sending the pre-encoded media to the SIP carrier. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] G729 Passthru?
I think you do need to buy the G729 for each call. If your system is using anything other than G729. True, you only need a CODEC if your system is using anything other than G729. In that case you would only be using pass through. Only if you are COding or DECoding do you need a CODEC. For G729 SIP phones, or G729 carriers asterisk will not do any coding or decoding and you do not need a G729 CODEC. A related issue is whether I can pre-encode recorded audio files with a G729 codec. So the server can send wakeup call messages to the SIP carrier without running the codec at call time, just sending the pre-encoded media to the SIP carrier. You should be able to, though I couldn't tell you how. Don Pobanz winmail.dat___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Billing Software
In reality, this is the one I've found that has exactly what our client needs. However, it seems to be a closed system so we are evaluating it further. AstBill and MOR don't seem to have the feature to offer referral credits out-of-the-box. Maybe we missed something? Thanks, Daniel -Original Message- From: Guillermo Salas M. [EMAIL PROTECTED] Sent: Sun, December 3, 2006 11:43 am To: [EMAIL PROTECTED] Subject: Re: [asterisk-users] Billing Software Have you found any solution ? I'm looking for the same product. Seems like astbill [1] and MOR [2] can manage reseller accounts. Regards, [1] www.astbill.com [2] www.kolmisoft.com On Thu, 2006-11-30 at 11:29 -0500, [EMAIL PROTECTED] wrote: We are looking for an offline billing solution. We have a couple of particular requirements: 1) Since it's offline, we need to be able to import the CDR. 2) A way to support account credits based on referrals. Meaning, that if a member refers a new account, that member would get a free month of service, or similar type credits. 3) Generate invoices in either HTML or PDF format so they can be printed or emailed to the actual customers. Does anyone know of a package that supports this? Would prefer open source. Thanks, Daniel ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Is there any Asterisk controllable thermostat?
I am wondering if there is any such thermostat available which can be controlled from Asterisk. Like you call your home pbx, dial some extension, e.g. 333 and it asks to set the temperature, you enter a temperature, and it sets the thermostat to that temperature. This thermostat will be very useful, e.g. when you're coming back home after a few days and now its snowing and you want home to be warm on your arrival, you can turn the furnace on an hour before your arrival. Is there any such thermostat available, and for that matter any other Asterisk controllable home automation devices? -- Zeeshan A Zakaria ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How can i processed with Call Snooping,
Hello Users, Nice to meet You, How can i Processed the call Snooping, it my fifth Requesting and posting to Users, Nobody replies it,,, in Call snooping , How can i record the Voip users, We can record to users by using Monitor Application , Can any give clear layout of the Snooping Feature... -- Thanks and Regards Ravi Prakash Sunkara [EMAIL PROTECTED] M:+91 9985077535 O:+91 40 23114549 F:+91 40 40208727 [EMAIL PROTECTED] www.hyperion-tech.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM01B installation
On Mon, Dec 04, 2006 at 09:12:21AM +0700, Mochamad Susantok wrote: Hi, iam new in this milis I have problem with TDM01B Installation, output zttool command is Unable to open /dev/zap/ctl: No such device or address and then i find the same IRQ uses VGA compatible controller and Communication controller is 169 What can i do next ? please your advice cat /proc/zaptel/* lspci modinfo | grep wctdm What version of Zaptel do you use? What distribution? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] HOW TO - Asterisk apps/modify and compile
hi all, i need to integrate and modify one of the application in asterisk/apps section... whenever i modified small steps ..in order to check and compile i 've to do recompile the whole asterisk module and it consuke to much time... please anyone couls you tell me, how can i modify it , compile and test the I/O in asterisk applications in a easy way... plz do reply .. Thanks for ur commands, Regards, nsthi ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] bristuff error: received SETUP message for callthat is not a new call
On Monday, November 27, 2006 10:23 AM Louis-David Mitterrand wrote: Hello, With the following setup: - asterisk 1.2.13, - zaptel 1.2.10 - bristuff 0.3.0-PRE-1v - quadbri card, Have you tried using bristuff 1v with the qozap driver of 1s? All qozap versions after 1s had serious problems (which seem to be fixed in soon to be released 1w). If this does not help, do a pri debug or better yet pri intense debug, describe the problem and contact the author with this info. Regards, JP ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] mwi for voicemail not showing up for realtime config.
Hello ppl, Am using realtime odbc storage for voicemail, sip users/peers, static for extensions and so on. My issue is I am not getting MWI for any fones, even tho I've got rtcachefriends=yes in sip.conf WIth tcpdump, I always see the NOTIFY going as Messages-Waiting:.no Voice-Message:.0/0.(0/0) even tho there are legitimate voicemails in the INBOX path for that particular users in the db. Any ideas, wot else shud i check for? TiA. cheerz - Ben. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is there any Asterisk controllable thermostat?
Never tried, but this should work: http://www.smarthome.com/3001.html Lots of neat stuff on that site. On 12/3/06, Zeeshan Zakaria [EMAIL PROTECTED] wrote: I am wondering if there is any such thermostat available which can be controlled from Asterisk. Like you call your home pbx, dial some extension, e.g. 333 and it asks to set the temperature, you enter a temperature, and it sets the thermostat to that temperature. This thermostat will be very useful, e.g. when you're coming back home after a few days and now its snowing and you want home to be warm on your arrival, you can turn the furnace on an hour before your arrival. Is there any such thermostat available, and for that matter any other Asterisk controllable home automation devices? -- Zeeshan A Zakaria ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM01B installation
On Mon, Dec 04, 2006 at 09:12:21AM +0700, Mochamad Susantok wrote: Hi, iam new in this milis I have problem with TDM01B Installation, output zttool command is Unable to open /dev/zap/ctl: No such device or address and then i find the same IRQ uses VGA compatible controller and Communication controller is 169 What can i do next ? please your advice cat /proc/zaptel/* No such file or directory lspci Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface modinfo | grep wctdm nothing ouptut What version of Zaptel do you use? What distribution? Debian 2.4 kernel asterisk-1.0.7 zaptel-1.0.7 -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - This email was sent using Student EEPIS-Webmail. http://student.eepis-its.edu/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
