[Asterisk-Users] gsm/wav format not recognized in Background() application
Hardware I have
Old celeron 300a asus p2b motherboard
256 megs ram
X100p
Esoniq pci 128 sound blaster
Basicly I have created sox rec and arecord gsm and wav files for a basic
start dial plan script and it does not seem to recognize the files I
created.
I can substitute any default sound file in /var/lib/asterisk/sounds
directory in place of my own custom Background(greeting) gsm file and it
will work when the Answer application kicks in. When I insert my own
sound files I hear nothing but a click. I can push in extentions and it
will revert to voicemail as requird.
Here is some of my extension.conf file
[incoming]
exten = s,1,Wait,1
exten = s,2,Answer()
exten = s,3,NoOp(${CALLERID})
exten = s,4,Background(greeting1)
exten = t,1,Goto(s,4)
exten = 100,1,Dial(IAX2/100)
exten = 100,2,Voicemail(u100)
exten = 200,1,Voicemail(u200)
I have created a new greeting.gsm called greeting1.
I can play is fine when using sox play command but for unknown reasons
not when run by asterisk. Any and all sound applications are not running
at the moment to prevent any conflics.
I created the file as a wav file like rec greeting1.wav -r 4000 vol 1
Then convert it withsox greeting1.wav -r 8000 -c 1 greeting1.gsm
resample -ql
I replay the gsm to test it. It sounds clear with some static.
Then these errors come up in CLI when dialing into my zap card:
NG[234]: format_wav.c:159 check_header: Unexpected freqency 4000
Jan 22 21:42:16 WARNING[234]: file.c:412 ast_filehelper: Unable to open
fd on /var/lib/asterisk/sounds /greeting1.wav
Jan 22 21:42:16 WARNING[234]: file.c:790 ast_streamfile: Unable to open
greeting1 (format unknown): No such file or
directory
Jan 22 21:42:16 WARNING[234]: pbx.c:4959 pbx_builtin_background:
ast_streamfile failed on Zap/1-1 for greeting1
Jan 22 21:42:26 WARNING[234]: format_wav.c:159 check_header: Unexpected
freqency 4000
Jan 22 21:42:26 WARNING[234]: file.c:412 ast_filehelper: Unable to open
fd on /var/lib/asterisk/sounds /greeting1.wav
Jan 22 21:42:26 WARNING[234]: file.c:790 ast_streamfile: Unable to open
greeting1 (format unknown): No such file or
directory
Jan 22 21:42:26 WARNING[234]: pbx.c:4959 pbx_builtin_background:
ast_streamfile failed on Zap/1-1 for greeting1
So anyone might give me a idea what I may be doing wrong?
here is my lsmod if this will help
James
Module Size Used byNot tainted
snd-pcm-oss37736 1 (autoclean)
snd-mixer-oss 12504 0 (autoclean) [snd-pcm-oss]
wcfxo 8384 1
zaptel175904 6 [wcfxo]
snd-ens1370 7780 2
snd-pcm56072 0 [snd-pcm-oss snd-ens1370]
snd-timer 13604 0 [snd-pcm]
snd-page-alloc 6328 0 [snd-ens1370 snd-pcm]
snd-rawmidi12740 0 [snd-ens1370]
snd-seq-device 3888 0 [snd-rawmidi]
snd-ak4531-codec4824 0 [snd-ens1370]
snd30852 1 [snd-pcm-oss snd-mixer-oss snd-ens1370
snd-pcm snd-timer snd-rawmidi snd-seq-device snd-ak4531-codec]
3c59x 25648 1
gameport1420 0 [snd-ens1370]
soundcore 3396 6 [snd]
agpgart43940 0 (unused)
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Re: [Asterisk-Users] Festival
On Mon, 2005-01-24 at 14:45, Gary wrote: On Mon, 24 Jan 2005 14:57:06 +1300, Matt Riddell wrote: Howard Lowndes wrote: Is it possible to get the Festival command to read the text from a system file rather than having it input as a text string? Is this a case of having to use AGI, or is there a simpler way? Most people would use AGI for that (combined with the text2wave or whatever program). In fact there may even be an example on the wiki. I might also add that if you look in the wiki for cepstral as well some good examples. And cepstral voices sound much nicer than festival :-) Never heard of it. Tks for the lead. . ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] gsm/wav format not recognized in Background() application
james wrote: Jan 22 21:42:26 WARNING[234]: format_wav.c:159 check_header: Unexpected freqency 4000 You might want to try encoding at 8000Hz instead of 4000Hz. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SPA-2000
Thanks for this info! :) This should be in the wiki, I couldn't find it on voip-info.org nor on ip-phone-forum.de Cheers! Remco On Sun, 23 Jan 2005, Chris Stenton wrote: 600 is for the US only. FXS impedence for UK 370+620||310nF Europe CTR21 270+750||150nF Chris - Original Message - From: Remco Barende [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, January 22, 2005 10:12 AM Subject: Re: [Asterisk-Users] SPA-2000 On Sat, 22 Jan 2005, Duane wrote: Remco Barende wrote: On Fri, 21 Jan 2005, Henry Devito wrote: Hi, I have not implemented any of the spa-2000's yet. Do they work ok with asterisk? Is the 2000 capable of having 2 FXS extensions off each one or is it two fxs ports with the same extension? They work pretty well, but I'm not impressed with the sound quality. Sound is quite soft and I have to adjust the input and output gains to something like +3 or +5 for in+out and then an annoying hiss is audible. I have a sipura 2000 and haven't had to alter gain at all, and no hiss, then again are you using ulaw or using g729? I use G711u. When I do not adjust the output gain the volume in+out is just too soft. Or would I need to change another setting? Under Regional I can also set port impedance. No idea though if the default value of 600 is ok for Europe? As for the original question, the 2 ports on the 2000 and the 3000 are both seperate SIP identities and you have to configure them as 2 seperate lines... ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Some more hardware and E1 questions
I will be using Debian, and as long as the Linux Kernel supports the SATA controller, the rest shouldn't be any problems. If it's SATA RAID, I probably will use ordinary Linux software RAID, since it's more powerful than the simple one in the controller. - Original Message - From: Leo Ann Boon [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, January 22, 2005 6:13 AM Subject: Re: [Asterisk-Users] Some more hardware and E1 questions Daniel Nyström wrote: Hi again folks! ;) As before, I will transform one E1 30 Channel PRI into 30 FXS channels using Adit 600. Now I'm into choosing server platform. And the two opponents are: * Dell PowerEdge 750 w/ SCSI RAID (or even SATA RAID1) * FujistuSiemens PRIMERGY RX100 S2 (SATA RAID1) If you're planning to use SATA RAID on PE750, make sure your Linux distro supports. Your best bet - use Redhat Enterprise Linux or one of it derivatives. I'm using Centos 3, it autodetects the RAID whilst Mandrake 10 failed. As I've seen people having problem with HP server, I havn't looked at it at all. What experience do you have with the alternatives above? Which would you recommend? And another question at the same time; what's really E1? How is E1 devices connected? Seems like regular Cat5 cables, but it problably ian't? If anyone's using Adit 600, did they send all cables required for connecting to the FXS channels? Seems like a very unique plug on the side of Adit. Thanks! BR Daniel Nyström ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 1.0.5
El 24/01/2005, a las 4:34, Russell Bryant escribió: Hello everyone, As you know, we released Asterisk 1.0.4 earlier this week. However, there was a callerid bug in chan_zap that has caused us to go ahead and make another release. Asterisk 1.0.5 is available at all of the usual locations. I'm sorry for any inconvenience this may cause. What about zaptel and libpri? are they ok, can continue running the versions i have now? Adrià Vidal ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] about call out : a strange question.
Hello all,
I want to use asterisk pbx to give a ring for sip user.when A call B ,
user B 's mobile will ring.(B always register his sip number and his mobile
number first.)
ignorepat = 9
exten = _9NXX,1,Dial(Zap/g2/${EXTEN:1})
exten = _9NXX,2,Congestion
but I want only let B's mobile ring,B can't access. or when B
access,the phone auto hang up.
what I should do?
who can told me?
thank you.
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[Asterisk-Users] Threeway calling
Can eny body tell me how to configure threeway calling using SIP channels? Thanks RegardsRitesh Jalan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Some more hardware and E1 questions
On Mon, 24 Jan 2005, Daniel Nyström wrote: Daniel Nyström wrote: And another question at the same time; what's really E1? How is E1 devices connected? Seems like regular Cat5 cables, but it problably ian't? If anyone's using Adit 600, did they send all cables required for connecting to the FXS channels? Seems like a very unique plug on the side of Adit. E1 is a serial interface with one of a few line encodings. Sometimes (often) a channel structrure is applied which leaves 31 channels of 64 kb/s each. Other protocols can then be applied on top of the E1 such as isdn etc. E1 is (almost) always deliverd as either an RJ45 plug with balanced 120 ohm impedance (most common) or as two coaxial cables with unbalanced 75 ohm impedance (uncommon). Other weird and wonderful connectors are sometimes used by specific equipment such as routers. For the rj45 case a normal cat5 cable will do. E1 uses the pairs 3-6 and 4-5. A cat5 cable normally has the pairs 1-2, 3-6, 4-5 and 7-8 of which ethernet uses 1-2 and 3-6. I have not looked at the Adit but I suspect they use the standard Amphenol connector that is very common in the telecom business. You can get a rj45 socket list with an amphenol connector on the back, or you can get a suitable pig-tail cable and punch it down into a punch-down block for furhter wireing. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAXy's apparantly failing in the field
Henry Devito wrote: Hi Matt, I was just getting ready to try to order a IP phone and ATA in the morning. This is the contact info I have. a.. email: [EMAIL PROTECTED] a.. Phone: +49 69 949 44 185 a.. Fax: +49 69 949 44 118 Thanks for the info, I also saw www.iaxtalk.com is advertising on -biz too (no i'm not affiliated with them, just nice to see more iax ata's and devices out there) matt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Auto callout - reminder - is it possible?
I'm trying to get a script working on a website to send out automatic email reminders to customers reminding them monthly to change furnace filters. I haven't got one running successfully, yet. That made me think - could it be done with a phone call using Asterisk? A monthly automated phone call to remind people to change their furnace filter? I have no ability to figure this out myself, but can it be done? Has it been done? Can I just search for an Asterisk application to do it and customize it for my own use? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] chan_skinny and firmware upgrade
Nothing to do with skinny, drop the new file(s) in your tftp directory and edit the .xml file to specify the new version, the phone will upgrade itself when it loads the config. Steve -Original Message- From: Subhi S Hashwa [mailto:[EMAIL PROTECTED] Sent: 23 January 2005 06:33 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] chan_skinny and firmware upgrade Hello all, I am trying to upgrade the firmware on my cisco 7910 without using CCM. I was told that chan skinny is possibly capable of doing that and would like to make sure. I have P00405000600 firmware which I have put in version in skinny.conf. the phone basiclaly stops at verifying load. tcpdump shows nothing happening apart from small amount of traffic to port 2000 (skinny). Does anyone have any ideas on how to get the new firmware into the phone? cisco instructions arent very helpful. PS unlike the bigger brother of the phone, this one does not request PS OS79XX.TXT file and is not SIP capable. -- Best regards, Subhi S Hashwa mailto:[EMAIL PROTECTED] When everything is heading your way, you're in the wrong lane. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The information contained in this email is intended for the personal and confidential use of the addressee only. It may also be privileged information. If you are not the intended recipient then you are hereby notified that you have received this document in error and that any review, distribution or copying of this document is strictly prohibited. If you have received this communication in error, please notify Brendata immediately on: +44 (0)1268 466100, or email '[EMAIL PROTECTED]' Brendata (UK) Ltd Nevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX UK Registered Office as above. Registered in England No. 2764339 See our current vacancies at www.brendata.co.uk ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Data calls with Asterisk
On Jan 24, 2005, at 1:50, Karim Mardhani wrote: I have about 10 remote locations which are collecting some data. I would like to upload that data every night. All remote locations have 56K modem. I was wondering can Asterisk be used to receive this data? Basically I will have an asterisk with 1 FXO card and have it receive data calls. Can asterisk receive data calls? Why use asterisk for that if you can simply plug a modem into the receiving computer and use mechansisms that are *made* for that purpose, such as PPP? jens --- Jens Vagelpohl [EMAIL PROTECTED] Software Engineer +49-(0)441-36 18 14 38 Zetwork GmbHhttp://www.zetwork.com/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Looking for a prepaid calling card platform
Hello Jim, we can offer you 100% working and fully tested Radius based billing system for Asterisk PBX with all this requirements except the credit card recharge support - but I think that building such a feature will not be a big problem :) We will need also some more detailed information about the multiple resellers support feature. If you are interested, please contact me offlist [EMAIL PROTECTED] Best regards, Lubo - AppRadius Project: Full RADIUS AAA support for Asterisk PBX http://appradius.minitelecom.org/ - James H. Thompson wrote: I'm looking for a prepaid calling card platform that: * easily scales to multiple servers with a common database for: redundancy, capacity, and performance Looking to start with capacity to handle 100 simultaneous calls and be able to easily scale to 1000+ simultaneous calls. * in addition to the normal anti-fraud measures, supports an API for easily adding new anti-fraud tests along the lines of the following: For each new call being attempted the system would invoke an external authentication program and pass: reseller ID, card ID, time left on card, called # and calling #; and the history/status for the last several calls including for each call the called #, calling #, call duration, call timestamp and call status (in-progress, completed, etc). Progrm would return: call OK, deny call with recording #x, invalidate card with recording #y. * ability to limit calls to a maxium duration and/or to require periodic IVR user response to continue a long call. * contolled, managed and provisioned with a web interface * support multiple resellers, each with password protected web access for managing their customers. * ability for customers to call an 800# to hear a recording giving them the user a local non-800 number they need to call to use the card. * credit card recharge support While willing to do minor customizations, would like to find something that is mostly install and go. Open source would be nice, but willing to pay for a well done package. Suggestions welcome. Thanks. Jim James H. Thompson [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk on sattelite link
marius baranescu wrote: Hi , I have a running Asterisk box . It is running great My problem is that I can not get connected to the world :) . Well, the sensible option then is to rent a cheap server somewhere with static IP and do VPN / Tunneling. My only option available here is a satellite connection. Ouch. It will work but there might be some lag in the conversations. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoIP Providers and Backbone Servers
On Jan 24, 2005, at 1:51, [EMAIL PROTECTED] wrote: Additionally, these small beginings enable people like myself to learn the industry quickly and get involved. It also allow us to learn about the Astrisk PBX system as well as the multitude of hardware and software that comprise this exciting field. You need to do what you have fun doing - anything else isn't worth doing. As long as you don't overrepresent yourself and/or customers end up being guinea pigs because your learning process has not proceeded far enough you have every right to work on that idea and make it happen. Even if you don't offer some flashy new feature others don't. I wish you good luck. jens --- Jens Vagelpohl [EMAIL PROTECTED] Zetwork GmbHhttp://www.zetwork.com/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk on sattelite link
Marius, I have * running in Houston, Texas and regularly run SIP from my office in Nigeria. We have our own earth station here and terminate in Canada and use the net from there to our data center. Here in Nigeria the phones are behind a PIX with NAT. From my experience, you should not have any problems. John Dunham -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of marius baranescu Sent: Monday, January 24, 2005 10:33 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Asterisk on sattelite link Hi , I have a running Asterisk box . It is running great My problem is that I can not get connected to the world :) . My only option available here is a satellite connection . I was testing different service providers but all of them are doing firewalling and NAT so SIP, IAX are not working I desperately need to get connected to the world :)) Please recommend me a good ISP for Middle East (permanent 2 way connection) , real IP adresses etc Best regards , Marius marius dot baranescu @ gmail dot com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Autio cut off at beginning of call
This is not normal; I do *not* have this issue with NuFone and I have placed a ton of calls through them daily for the past year. I don't recall having this problem with voicepulse connect when I used them, nor do I have the issue with iax.cc for inbound calls. I'm experiencing this on two separate * systems. The symptoms appear only on outbound calls, never inbound. I think it's important to note that this affects outbound calls made either of SIP or IAX, and through multiple providers. It very much sounds like it's something on your end... How about some specifics? I'm using the defaults found in the [general] section of iax.conf and sip.conf. I'm using Asterisk version 1.0.3, but I've experienced this problem with every version I've used over the past year. It also does not matter if the call is placed from a SIP phone or an FXS channel. What additional info would be most helpful? Thanks, Reid ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] UPS for Asterisk
I have several Linux machines some running on really old hardware and some on brand new, some run old distros (RedHat 6) and some new (FC3 or CentOS). All of them experienced power failure more then once, none of them has failed to load after a reboot. BUT, Asterisk is running your PBX. Your PBX isn't your proxy server, it isn't your web server, mail server, firewall, or whatever you're used to run on linux. Even though it would seem that down time on all of these production machines is bad, these are all systems that have no counter part in the legacy world, and that we all agree may have some downtime along the road. On the other hand, telephony down time is unacceptable. PBXs have a counter part. Plain old PBXs are expected to run 24x7. real 24x7, with uptimes of 99.999. And if you think about it, they actually do. So people will expect your asterisk installation to do the same. Besides, when a mail server goes down for ten minutes, when it comes back up you still get your mail. This is not true for your PBX. Our asterisk installation has software RAID, has a UPS, has recover CDs burnt and ready to be used (http://www.builderau.com.au/architect/sdi/0,39024602,20269582,00.htm) And still, my knees are shaking. In short, GET 100$ and BUY A UPS. It's worth it. -Original Message- From: Nick Bachmann [mailto:[EMAIL PROTECTED] Sent: Monday, January 24, 2005 5:30 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] UPS for Asterisk Andrew Kohlsmith wrote: On January 23, 2005 04:04 pm, Mike Sander wrote: Is the harddisk activity on a standard asterisk install such that I don't really have to worry if the power cuts?? Not typically; there isn't much writing going on, this is true. Are you that cash strapped that a $75 UPS with a serial port is out of your budget? No kidding... the cost of a server than won't come up again is much more substantial than the countermeasure... the $75 (you can get a 350 Va for $45 even!) and a slightly less energy efficient system. If you can afford to spend more, a decent active UPS would keep your power conditioned as well... As I understand, if HD activity is minimal, the probability of HD failure is significantly reduced. HDDs don't fail because they lose power. Unless the heads crash, which can happen if power fails. I know HDD manufacturers have done head unloading and such recently, but the risk is still higher if power is suddenly lost during a write. Nick ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. MailScanner thanks transtec Computers for their support. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] PSTN and Asterisk
Hi quys, I look for a solution for interconnection beetwen PSTN and VoIP. My application have to treat few protocols comming from PSTN lines and mixing data , dtmf and voice. Can I use Asterisk for : PSTN-- Asterisk (converting analog call to IP) -- MyApplication( translation protocolsand do some workswith incomming data) What hardware I can use for this? Do use Asterisk G.711 protocol? Thanks Vassili ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Autio cut off at beginning of call
I'm using the defaults found in the [general] section of iax.conf and sip.conf. I'm using Asterisk version 1.0.3, but I've experienced this problem with every version I've used over the past year. It also does not matter if the call is placed from a SIP phone or an FXS channel. What additional info would be most helpful? Check the load on your server(s). Regards, Senad Jordanovic Bicom Systems www.bicomsystems.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Autio cut off at beginning of call
On Jan 24, 2005, at 11:27, Senad Jordanovic wrote: Check the load on your server(s). I have the same problem with calls to and from NuFone. It's probably not load-related because the load is non-existent on that box. It runs nothing but Asterisk with a very simple network-only config where no telephony hardware is used. The only thing connected to it is an IAXy with a cordless hanging off it. jens P.S.: Am I the only happy IAXy user out there or what? I love that thing. Never any trouble. ;) --- Jens Vagelpohl [EMAIL PROTECTED] Zetwork GmbHhttp://www.zetwork.com/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Zapata in Australia
Does anybody what the regional settings are to use an x100p (clone) card with Asterisk in Australia? I got mine installed and recognised by * but I get no sound and terrible hangup detection. Basically after each test call to the landine number (plugged into the x100p card) I need to unplug the cord and plug it back in to get a normal dialtone. When * answers the call (or diverts it to any internal IP phone) there is absolutely no sound. many thanks manny ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Autio cut off at beginning of call
Check the load on your server(s). Load average is always at or near 0. This is on a dedicated machine doing nothing but routing calls. No voicemail, music on hold, etc. I noticed something in a packet capture that may or may not be significant. When I place a call, the capture shows about 3 seconds of audio data before the IAX ANSWER packet. Could this be a symptom of the problem? Below is a packet dump if an outbound IAX call made from a Zaptel FXS channel. No. TimeSourceDestination Protocol Info 9 11.987842 asterisk 66.234.228.170IAX2 IAX, source call# 4, timestamp 3ms REGREQ 11 12.086365 66.234.228.170asterisk IAX2 IAX, source call# 356, timestamp 4ms REGACK 12 12.086456 asterisk 66.234.228.170IAX2 IAX, source call# 4, timestamp 4ms ACK 13 12.158422 asterisk 66.234.228.160IAX2 IAX, source call# 6, timestamp 14ms NEW 14 12.258421 66.234.228.160asterisk IAX2 IAX, source call# 228, timestamp 10ms AUTHREQ 15 12.258514 asterisk 66.234.228.160IAX2 IAX, source call# 6, timestamp 114ms AUTHREP 16 12.383395 66.234.228.160asterisk IAX2 IAX, source call# 228, timestamp 109ms ACCEPT 17 12.383552 asterisk 66.234.228.160IAX2 IAX, source call# 6, timestamp 109ms ACK 18 12.397652 asterisk 66.234.228.160IAX2 Voice, source call# 6, timestamp 253ms, Raw mu-law data (G.711) 19 12.417636 asterisk 66.234.228.160IAX2 Mini packet, source call# 6, timestamp 273ms, Raw mu-law data (G.711) 20 12.437635 asterisk 66.234.228.160IAX2 Mini packet, source call# 6, timestamp 293ms, Raw mu-law data (G.711) 21 12.457634 asterisk 66.234.228.160IAX2 Mini packet, source call# 6, timestamp 313ms, Raw mu-law data (G.711) 22 12.477634 asterisk 66.234.228.160IAX2 Mini packet, source call# 6, timestamp 333ms, Raw mu-law data (G.711) 23 12.497634 asterisk 66.234.228.160IAX2 Mini packet, source call# 6, timestamp 353ms, Raw mu-law data (G.711) 24 12.498398 66.234.228.160asterisk IAX2 IAX, source call# 228, timestamp 253ms ACK 25 12.515119 66.234.228.160asterisk IAX2 Control, source call# 228, timestamp 112ms stop sounds 26 12.515161 asterisk 66.234.228.160IAX2 IAX, source call# 6, timestamp 112ms ACK 27 12.517701 asterisk 66.234.228.160IAX2 Mini packet, source call# 6, timestamp 373ms, Raw mu-law data (G.711) 28 12.537634 asterisk 66.234.228.160IAX2 Mini packet, source call# 6, timestamp 393ms, Raw mu-law data (G.711) 29 12.557634 asterisk 66.234.228.160IAX2 Mini packet, source call# 6, timestamp 413ms, Raw mu-law data (G.711) 30 12.565685 66.234.228.160asterisk IAX2 Control, source call# 228, timestamp 115ms unknown (0x0e) 31 12.565732 asterisk 66.234.228.160IAX2 IAX, source call# 6, timestamp 115ms ACK 32 12.577636 asterisk 66.234.228.160IAX2 Mini packet, source call# 6, timestamp 433ms, Raw mu-law data (G.711) 33 12.585428 66.234.228.160asterisk IAX2 Voice, source call# 228, timestamp 20ms, Raw mu-law data (G.711) 34 12.585505 asterisk 66.234.228.160IAX2 IAX, source call# 6, timestamp 20ms ACK 35 12.597633 asterisk 66.234.228.160IAX2 Mini packet, source call# 6, timestamp 453ms, Raw mu-law data (G.711) 36 12.606341 66.234.228.160asterisk IAX2 Mini packet, source call# 228, timestamp 40ms, Raw mu-law data (G.711) 37 12.617633 asterisk 66.234.228.160IAX2 Mini packet, source call# 6, timestamp 473ms, Raw mu-law data (G.711) . . . Nothing of interest in here, just audio data . . 327 15.477620 asterisk 66.234.228.160IAX2 Mini packet, source call# 6, timestamp ms, Raw mu-law data (G.711) 328 15.486797 66.234.228.160asterisk IAX2 Mini packet, source call# 228, timestamp 2920ms, Raw mu-law data (G.711) ---329 15.487713 66.234.228.160asterisk IAX2 Control, source call# 228, timestamp 2923ms ANSWER 330 15.487734 asterisk 66.234.228.160IAX2 IAX, source call# 6, timestamp 2923ms ACK 331 15.497627 asterisk 66.234.228.160IAX2 Mini packet, source call# 6, timestamp 3353ms, Raw mu-law data (G.711) 332 15.517623 asterisk 66.234.228.160IAX2 Mini packet, source call# 6, timestamp 3373ms, Raw mu-law data (G.711) 333
[Asterisk-Users] OT: Libnewt sourcecode?
Hi, I'm trying to compile zttool from the Zaptel lib, but I just can't find the sorcecode for Libnewt. Anyone got a link? Since i'm using LFS, I can't use precompiled packages. -- Med venlig hilsen / Best regards Michael Løjtnant - Systems Engineer ZyXEL Communications A/S Columbusvej 5 - 2860 Søborg Tel (+45) 3955 0700 - Fax (+45) 3955 0707 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Mediatrix voip gateway 1124 and 1204 in UK setting
Hello! We are located in the UK, and we are planning to replace our old pbx with an asterisk based pbx. For outgoing calls our present pbx is connected to three PSTN lines which all have the same number. Internally, the pbx caters for quite a few extensions, and each extension can make outbound phone calls. Only very rarely does our call volume exceed three simultaneous connections (inside to inside plus inside to outside). We have looked into the issue of connecting the phones and the outside lines to the system. For the fxo connectivity we want to stick with the three PSTN lines, because they worked for us and we don't see a need to upgrade to ISDN. The asterisk system will be also connected to the internet anyway so we can perform VOIP calls. For the fxs connectivity we want to re-use the old telephone wiring and provide standard two-wire telephones. Putting in IP phones would mean a massive installation effort, as we would have to put an entire new computer network in place - plus many IP phones constantly connected to mains, plus admin headaches, plus security issues and so on. The two wire solution seems the best solution for our setting. We have looked into using a channel bank for the analog conectivity, and we are currently in contact with Carrier Access to purchase a new Adit 600 unit with space for 48 extensions. We cannot provide fxo connectivity via the channel bank because the fxo card from CA seems not to be EU approved. One downside of the channel bank is that we need a special T1 card for it to operate with the asterisk pbx. Also, channel banks seems to be a particular US concept, so we would have difficulties to get replacement parts, if something breaks. Recently I heard of the alternative solution of a voip gateway, and the particular units I have seen are the Mediatrix 1124 for fxs connection and the Mediatrix 1204 for the fxo connection. Both units support the SIP protocol, so it should be possible to connect them to the asterisk PC via standard network connection. Mediatrix seems to have resellers in Europe as well, so it might be possible that their devices are Europe approved as well. Question: * Does anyone have any experience with these units in a UK setting? * For the 1124: Does it work with standard UK two wire phones? Are there impedance problems (especially concerning echo problems)? Is the audio quality sufficient? Are they transparent to the asterisk system, i.e. does each fxs port look like a separate IP phone to the asterisk system? * For the 1204: Would it be approved for connection into the UK PSTN (The prospectus from Mediatrix didn't say anything about regulatory approvals)? Can they initiate outside calls / receive incoming calls or are there problems (signalling compatible with UK PSTN)? Are they transparent to the asterisk system, i.e.does each fxo port look like a separate IP phone to the asterisk system? I do realize that these questions are quite broad, but do appreciate any info. Thank you very much for your consideration. -- dyslexics of the world - untie ! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] OT: Libnewt sourcecode?
Michael, you need the package 'newt-devel'. See also http://lists.digium.com/pipermail/asterisk-users/2003-May/011185.html for further reference. Happy asterisking! Peter -- dyslexics of the world - untie ! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk starting problem
Hi, I have a little problem running Asterisk. I just got the asterisk, zapttel and libpri sources from cvs. I built and installed it. Next I installed the sample configuration. The problem arise when I try to start Asterisk. Running asterisk -c I get the following error [chan_phone.so] = (Linux Telephony API Support) == Parsing '/etc/asterisk/phone.conf': Found == Registered channel type 'Phone' (Standard Linux Telephony API Driver) Jan 24 12:25:01 WARNING[1625]: chan_oss.c:241 sound_thread: Read error on sound device: Resource temporarily unavailable [chan_h323.so]Jan 24 12:25:01 WARNING[1625]: loader.c:302 __load_resource: /usr/lib/asterisk/modules/chan_h323.so: undefined symbol: ast_pthread_create Jan 24 12:25:01 WARNING[1625]: loader.c:510 load_modules: Loading module chan_h323.so failed! Note that I'm running Asterisk on my own laptop without any boards. I'm running Gnu\Linux with Debain Sarge distro. Before I installed Asterisk using apt-get install asterisk and it worked fine. Any ideas??? many thanks Stefano. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoIP software for MAC OS older than X?
There are a few products for OS9 but most of the development I've seen is for Mac OSX. Depending upon what features your idea of a VoIP soft client must implement you could try sipc from Columbia University (actually a spinoff), Session from wave3software and possibly Xten but I forget how far back Xten Mac development goes. Daiku wrote: Hello, is there anybody reading this who has experience with VoIP (IAX or not) on Macintosh computers? If so, have you ever seen or heard of (even an experimental, i.e., never marketed) VoIP application for any of the older Mac OSs, such as 9, 8, or 7? I can't quite believe that VoIP is such a recent idea that it was invented only *after* Mac OS X had become firmly established, but so far my searches have turned out nothing. However, not all good stuff and good ideas are on the web,so a community of knowledgable people often has information that a web search cannot produce. Appreciate any leads and comments... Thanks: H.D. -- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoIP software for MAC OS older than X?
is there anybody reading this who has experience with VoIP (IAX or not) on Macintosh computers? If so, have you ever seen or heard of (even an experimental, i.e., never marketed) VoIP application for any of the older Mac OSs, such as 9, 8, or 7? I can't quite believe that VoIP is such a recent idea that it was invented only *after* Mac OS X had become firmly established, but so far my searches have turned out nothing. However, not all good stuff and good ideas are on the web,so a community of knowledgable people often has information that a web search cannot produce. The old MacOS' lack of scheduling and other fundamental functionality makes it not worth programming real-time apps on. So don't ask. Kick it out, use ATAs or IP phones or a PC. There's no point trying. You'll end up doing something else in the end anyway. roy ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoIP software for MAC OS older than X?
Hello Steve, thanks for the info...! I checked out those vendors' sites but did not find anything for a Mac OS before OS X at this time - but i have noticed with other software vendors that many of them tend to remove slightly older versions from their sites, so no surprise here... ;-) Regards: H.D. Quoting from message: 05/01/24 20:31 +0900 sent by Steve Blair: [...] sipc from Columbia University (actually a spinoff), Session from wave3software and possibly Xten [...] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] zaphfc no callerid incoming to SIP phone butvisible in logfile
Try commenting out the line
pritrustusercid = yes
Or set it to 'no'.
That worked for me.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jens
Sent: Friday, January 21, 2005 7:18 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] zaphfc no callerid incoming to SIP phone
butvisible in logfile
Hi,
I think you didn't set usecallerid=yes in your zapata.conf?
Another way is to set the callerid in your extensions.conf via exten =
807440,2,SetCIDNum(0${CALLERIDNUM}). So you also have a 0 in front of the
displayed number - nice for callback.
regards
Jens
Hello,
I've added a ZAPHFC card to my CAPI based system. Calls coming in via
ZAPHFC do not forward the caller id to the SIP phones. Calls coming in
via CAPI do forward the caller id to the SIP phones.
--
Jens Lentföhr
http://www.jens-it.de
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Re: [Asterisk-Users] Power Alarm Error - Help
On Sun, Jan 23, 2005 at 11:56:50AM -0600, Michael K. Rodriguez User wrote: I had a similar problem with power. I connected Asterisk to a Belkin UPS 1200VA and the the server would boot up and asterisk would load but the T1s on the Quad T1 card failed to come up. I placed a loop on the card and still no change. Finally, I removed the UPS and the T1s came up. Do know if this will help you, but the T1 card seems to be delicate with power. I've been having similar trouble with one of my units. I put a UPS on the system and it seemed to get better, but how that module fails regularly. Incidentally, when a module dies and holds the circuit open, I can top asterisk, unload the kernel modules, reload them, run ztcfg, and restart * w/o restarting the system. However, it does require * to stop and that is annoying. I moved my problematic phone to a different fxs module and all seems fine. We have a similar problem with a TDM/FXS module in a different location. I've written digium support, but they are kinda slow in responding. On 1/23/05 10:31 AM, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: I have been getting the following message in Asterisk and it shuts Asterisk down, needing a reboot. Power alarm on Module 2 I have (1) TDM400P with (2) FXS (2) FXO cards (1) X100P card Any ideas? Since nobody answered, I'll guess something :) Did you plug the power on the TDM400P ? since you have FXS ports, you need to plug it in ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- [This E-mail scanned for viruses by Declude Virus] -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DIAX 0.9.9g more features and higher stability
Em Sáb 22 Jan 2005 07:51, Dan escreveu: Hi all, There is someone on this list having latency issues with DIAX who can do this trace? I'm not able to dupplicate this behaviour here and as I'm behind a NAT I cannot use 2 DIAX phones connected to an external Asterisk server (or there is a workaround for this?). Hi Dan. I could help on it, but I'll be able to get this trace only on wednesday 26... Tks. -- D e n i s G a l v ã o iSolve - Solve Is Our Business Av. Candido de Abreu, 526 1206B CEP: 80530-000 - Curitiba - PR +55 41 252-2977 http://www.isolve.com.br ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Not answering PSTN until SIP answers
Hi, I was just wondering whether or not anybody has a dial plan or some advice on getting a SIP phone to ring without answering the PSTN line so that the caller doesn't have to pay for the phone call unless it actually get answered by a human or the answering machine after 40 seconds. I had a look through the wiki but there wasn't anything I could find (probably the wrong search terms). Any advice is greatly appreciated. Kind Regards Stuart ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Auto callout - reminder - is it possible?
Check sample.call in the asterisk tarball... edit the file, move it to /var/spool/asterisk/outgoing and it'll dial and connect de callee with the extension of your choice... Greetings On Mon, 24 Jan 2005 02:57:13 -0600, Roger Hanson [EMAIL PROTECTED] wrote: I'm trying to get a script working on a website to send out automatic email reminders to customers reminding them monthly to change furnace filters. I haven't got one running successfully, yet. That made me think - could it be done with a phone call using Asterisk? A monthly automated phone call to remind people to change their furnace filter? I have no ability to figure this out myself, but can it be done? Has it been done? Can I just search for an Asterisk application to do it and customize it for my own use? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: Libnewt sourcecode? - Found it :-)
After some alternative seaching, I finally found a source that worked: http://devel.santafelinux.com/source/newt-0.51.6/ Perhaps someone could add the link to the wiki? ./Michael On Mon, 24 Jan 2005 12:04:20 +0100 Michael Løjtnant [EMAIL PROTECTED] wrote: Hi, I'm trying to compile zttool from the Zaptel lib, but I just can't find the sorcecode for Libnewt. Anyone got a link? Since i'm using LFS, I can't use precompiled packages. -- Med venlig hilsen / Best regards Michael Løjtnant - Systems Engineer ZyXEL Communications A/S Columbusvej 5 - 2860 Søborg Tel (+45) 3955 0700 - Fax (+45) 3955 0707 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Med venlig hilsen / Best regards Michael Løjtnant - Systems Engineer ZyXEL Communications A/S Columbusvej 5 - 2860 Søborg Tel (+45) 3955 0700 - Fax (+45) 3955 0707 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Not answering PSTN until SIP answers
I was just wondering whether or not anybody has a dial plan or some advice on getting a SIP phone to ring without answering the PSTN line so that the caller doesn't have to pay for the phone call unless it actually get answered by a human or the answering machine after 40 seconds. I had a look through the wiki but there wasn't anything I could find (probably the wrong search terms). Any advice is greatly appreciated. The default operation for an incoming zap call ringing a sip phone is to not answer the call until the sip phone is picked up. That implies you've got something in your dialplan that is telling the zap interface to answer the incoming call right away. You might post your relavent sections of zapata.conf and extensions.conf if you want someone to help. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Not answering PSTN until SIP answers
This is easy as pie. Just Dial without an Answer. When the device you specified in dial picks up, it will pick up the line. Done and done. On Mon, 24 Jan 2005 20:40:04 +0800, Stuart Elvish [EMAIL PROTECTED] wrote: Hi, I was just wondering whether or not anybody has a dial plan or some advice on getting a SIP phone to ring without answering the PSTN line so that the caller doesn't have to pay for the phone call unless it actually get answered by a human or the answering machine after 40 seconds. I had a look through the wiki but there wasn't anything I could find (probably the wrong search terms). Any advice is greatly appreciated. Kind Regards Stuart ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Is it something someone said, was it something someone said? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Not answering PSTN until SIP answers
Dear Jon and Rich, thank you very much for that. I will give it a shot. I do have an Answer() in the sequence which will be the problem. Sort of turned out to be a dumb question didn't it maybe I should play on the edge and take a few more risks when experimenting. Again thank you very much. Regards Stuart On Monday, Jan 24, 2005, at 21:08 Australia/Perth, Jon Radon wrote: This is easy as pie. Just Dial without an Answer. When the device you specified in dial picks up, it will pick up the line. Done and done. On Mon, 24 Jan 2005 20:40:04 +0800, Stuart Elvish [EMAIL PROTECTED] wrote: Hi, I was just wondering whether or not anybody has a dial plan or some advice on getting a SIP phone to ring without answering the PSTN line so that the caller doesn't have to pay for the phone call unless it actually get answered by a human or the answering machine after 40 seconds. I had a look through the wiki but there wasn't anything I could find (probably the wrong search terms). Any advice is greatly appreciated. Kind Regards Stuart ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Is it something someone said, was it something someone said? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users !DSPAM:41f4f3df86395883018142! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Power Alarm Error - Help
I had a similar problem with power. I connected Asterisk to a Belkin UPS 1200VA and the the server would boot up and asterisk would load but the T1s on the Quad T1 card failed to come up. I placed a loop on the card and still no change. Finally, I removed the UPS and the T1s came up. Do know if this will help you, but the T1 card seems to be delicate with power. I've been having similar trouble with one of my units. I put a UPS on the system and it seemed to get better, but how that module fails regularly. Incidentally, when a module dies and holds the circuit open, I can top asterisk, unload the kernel modules, reload them, run ztcfg, and restart * w/o restarting the system. However, it does require * to stop and that is annoying. I moved my problematic phone to a different fxs module and all seems fine. We have a similar problem with a TDM/FXS module in a different location. I've written digium support, but they are kinda slow in responding. It sounds like the same problem that many of us have seen over the last several months. Digium support has been trying to identify the root cause and has been very quiet in terms of comments, etc. Replacing the little fxo/fxs module has apparently corrected the problem in some cases, but its fairly obvious the tdm card still has a problem that has not yet been diagnosed/fixed. Dropping/restarting the drivers is the only known fix (bypass) thus far. If you look back over the previous posts relative to tdm problems, you'll see lots of comments about ups's, power supplies, pstn lines, etc, etc. The majority of these have indicated the problem takes a fair amount of time (eg, days or weeks) before recurring, suggesting the problem might be related to a memory leak, temperature, or some other cause that involves 'time'. Shutting down an * system to install a ups, etc, disturbs just about everything associated with the intermitant problem. Jumping to a conclusion that a ups (etc) fixed a problem is essentially ignoring all of the other disruptions that happened during the ups install. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Threeway callin
Can eny body tell me how to configure threeway calling using SIP channels? Thanks RegardsRitesh ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Not answering PSTN until SIP answers
Cool, is this equal if you use ringer groups or queues? At home it's a bit silly to create a call queue that is answering and put a caller in queue. I just want several phones to ring and only answer the phone when somebody is picking it up. I tried just specifying several sip devices to ring but if one of all the sip phones specified is not registered any call will immediately be forwarded to voicemail. Thanks! On Mon, 24 Jan 2005, Jon Radon wrote: This is easy as pie. Just Dial without an Answer. When the device you specified in dial picks up, it will pick up the line. Done and done. On Mon, 24 Jan 2005 20:40:04 +0800, Stuart Elvish [EMAIL PROTECTED] wrote: Hi, I was just wondering whether or not anybody has a dial plan or some advice on getting a SIP phone to ring without answering the PSTN line so that the caller doesn't have to pay for the phone call unless it actually get answered by a human or the answering machine after 40 seconds. I had a look through the wiki but there wasn't anything I could find (probably the wrong search terms). Any advice is greatly appreciated. Kind Regards Stuart ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoIP software for MAC OS older than X?
Hi Roy, thanks for the comments... Quoting from message: 05/01/24 20:50 +0900 sent by Roy Sigurd Karlsbakk: The old MacOS' lack of scheduling and other fundamental functionality makes it not worth programming real-time apps on. So don't ask. Kick it out, use ATAs or IP phones or a PC. There's no point trying. You'll end up doing something else in the end anyway. Very much agreed on the last point: i'll most certainly end up using something else in the end, and that will either be an IP phone or, if it turns out to be working well enough, the IAXy plus a regular phone. A hardware based solution will be much better in my case, since it means portability (i travel a lot) and i won't need a computer or headset to make phone calls. Thanks regards: H.D. PS: In the meantime i am just playing (i.e., learning something), and old software is part of that. By the way, the e-mail software i am using right now (Eudora 1.4.3) seems to have been released 12 years ago (would that be the bronze age, in computer terms?)... ;-) -- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Auto callout - reminder - is it possible?
- Original Message - From: Julian J. M. [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, January 24, 2005 6:46 AM Subject: Re: [Asterisk-Users] Auto callout - reminder - is it possible? Check sample.call in the asterisk tarball... edit the file, move it to /var/spool/asterisk/outgoing and it'll dial and connect de callee with the extension of your choice... Greetings On Mon, 24 Jan 2005 02:57:13 -0600, Roger Hanson [EMAIL PROTECTED] wrote: I'm trying to get a script working on a website to send out automatic email reminders to customers reminding them monthly to change furnace filters. I haven't got one running successfully, yet. That made me think - could it be done with a phone call using Asterisk? A monthly automated phone call to remind people to change their furnace filter? I did see the wiki items: asterisk auto-dial out deliver message and Asterisk Auto-dial out and think I may be able to muddle my way through getting that working (although that may be questionable) but is it feasable to integrate this wit a website, where a user enters a phone number in a form, then asterisk somehow gets this information from the website and it gets added to a database or some way for Asterisk to gather that information and make the phone calls automatically? Maybe with another option on what type of notification it would be (furnace filter, 1 month reminder, 2 month reminder) based on a variable in the form? Meanwhile, I'll work on at least getting the auto dialout with playing a pre-recorded file working. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DIAX 0.9.9g more features and higher stability
Hi Denis, - Original Message - From: Denis Galvão - iSolve [EMAIL PROTECTED] Em Sáb 22 Jan 2005 07:51, Dan escreveu: Hi all, There is someone on this list having latency issues with DIAX who can do this trace? I'm not able to dupplicate this behaviour here and as I'm behind a NAT I cannot use 2 DIAX phones connected to an external Asterisk server (or there is a workaround for this?). I could help on it, but I'll be able to get this trace only on wednesday 26... I have send one trace of such a call to Steve to further debug iaxclient library. Thanks a lot, Dan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Any experience with Sangoma cards?
Hello, Yes. I've had good experience with all three you mentioned. Jon - Original Message - From: Robert Augustyn [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Sunday, January 23, 2005 2:17 PM Subject: [Asterisk-Users] Any experience with Sangoma cards? Hi, I am considering A101/102/104 cards for my asterisk installations. Has any of you used these or any Sangoma cards in such environment? Any thoughts? How do they stack up against Digium cards? Any input would be greatly appreciated. robert ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: VoIP software for MAC OS older than X?
Roy Sigurd Karlsbakk [EMAIL PROTECTED] writes: The old MacOS' lack of scheduling and other fundamental functionality makes it not worth programming real-time apps on. So don't ask. Kick it out, use ATAs or IP phones or a PC. ...or install a Unix on the Mac. :-) -tih -- Tom Ivar Helbekkmo, Senior System Administrator, EUnet Norway Hosting www.eunet.no T +47-22092958 M +47-93013940 F +47-22092901 FWD 484145 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DTMF issues (handytone)
morn all, I ran into a strange issue last night and have not been able to find resolution either in documentation (wiki) or experamentation. Using a handyton to feed dial tone to a pbx I am able to connect both ways with no problem. If I make a call from the pbx through asterisk I can sent DTMF tones with no problem. If I call into the pbx (through) the Handytone I am not able to pass DTMF tones. I hear a slight beep then silence. I have tried the Handytone set for DTMF info and rfc-2833 (as well as exp with inband) as well as the sip.conf entry for it. I thought I had a pretty good handle on how DTMF was intercepted and regenerated but ... ahhem .. guess not. Thanks for any Ideas in advance. mike -- |- - - - - - - - - - - - - - - - - - - -| |-Mike Deweyof -| |= All Technologies Unlimited, Inc =| |- phone: 303.667.0357 -| |- e-mail: [EMAIL PROTECTED] -| ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Auto callout - reminder - is it possible?
On January 24, 2005 08:38 am, Roger Hanson wrote: I did see the wiki items: asterisk auto-dial out deliver message and Asterisk Auto-dial out and think I may be able to muddle my way through getting that working (although that may be questionable) but is it feasable to integrate this wit a website, where a user enters a phone number in a form, then asterisk somehow gets this information from the website and it gets added to a database or some way for Asterisk to gather that information and make the phone calls automatically? Maybe with another option on what type of notification it would be (furnace filter, 1 month reminder, 2 month reminder) based on a variable in the form? The callout in the wiki blows goatass -- at least the voicemail one -- I have a cron script which looks for messages in the watched INBOX folders and generates a callout every 5 minutes until the message is listened to (at which time it is moved to the Old folder). As far as integrating with a website or database -- that is a piece of cake. Your backend logic just determines when a call is needed and gerates the approprate .call file. Just remember to create it in /tmp or something, close it and then MOVE it to the outgoing spool instead of creating and working on it in the outgoing spool. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voicemail folders
Title: Message Hi, How can I rename existing voicemail folders (INBOX - Inbox; Old - Archive)? Regards, Stojan Sljivic ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sipura Behind NAT howto
I am trying to get a SPA-3000 to work behind NAT - for the sake of the exercice. The SPA is on the local network at the address 192.168.0.125 behind a NATted linux router. The machine I am trying to work with is a friend's (let's call it lolo.dyndns.org) and I've installed Asterisk 1.0.3 on it. I can see the SPA register but when I try to make an outbound call I get the message: Jan 24 14:35:03 NOTICE[3184]: chan_sip.c:7295 handle_request: Unable to create/find channel Jan 24 14:35:03 NOTICE[3184]: chan_sip.c:7295 handle_request: Unable to create/find channel Jan 24 14:35:04 NOTICE[3184]: chan_sip.c:7295 handle_request: Unable to create/find channel Also, when I comment out the 'secret=' line from sip.conf, everything seems to work just fine... Any ideas what's going on? Cheers, Jean-Michel. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Best VPN server for * and woad warriors using windows?
I would like to install vpn software on the * server for roadwarriors to connect to with laptops running windows. OK, take a hard look at this before you get too far. Installing VPN software *on* the Asterisk box is not a good idea. Now, you haven't explained the volume of users on the box, or the availability needs of the box, but either way, this is bad practice. The term roadwarriors' makes me think this is for a business. Actually, both. I want to use it when I on holiday (then I am the roadwarrior) and want to make a cheap call home but I'm also considering it for business use. Why is it bad to put a vpn server on the * box? I will not have any users logging into it or anything and none of the users will have shell on the box. Also that particular VPN connection will not be used for anything else but phonecalls. There are numerous vpn server daemons around and I found many messages about some of them using tcp/udp etc and instead of trying them all out hopefully someone can recommend one? If you want IPSec, take a look at OpenWall. If you must run this on your asterisk box, so be it. I was considering IPSEC because I heard it is safer or more secure than PPTP and Windows XP supports IPSEC natively (or so it claims). I don't care about connecting Win9x clients, stoneage hardware shouldn't be doing voip anyways. :) Thanks!! Remco ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] DTMF issues (handytone)
Hi Mike, I have tried the Handytone set for DTMF info and rfc-2833 (as well as exp with inband) as well as the sip.conf entry for it. From my experience DTMF with any Grandstram device works well only with SIP INFO method ... give it a try (remember to set it up on asterisk as well). Best regards, Ivan Meic ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to display number being dialed
Hi, I have setup asterisk setup using FXO card with four ports. As every thing is working fine and I have used analog phones for calls, I also have purchased different numbers, when some one dials my numbers the caller number is displayed on analog phone, can any one of you will help to guide me so that when some dials my asterisk number for example xxx instead of caller number my umber (which is dialed by client should be displayed) xxx should be displayed . A quick response in this regard will be highly appreciated Regards, Mazhar Nettechltd.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Not answering PSTN until SIP answers
Dear Remco,
It now works for me with several SIP devices in a ring group without
answering until one of them answers. I think (from memory) that this is
the code that you need:
In extensions.conf setup a virtual phone like this
PHONES4=SIP/102SIP/103SIP/104
and define the voicemail box if required, for example a communal
voicemail box 1
PHONES4VM=1
Then in your incoming caller context, DO NOT put Answer, the first
priority should be Dial
exten = s,1,Dial(${PHONES1},40,tr)
exten = s,2,Macro(vmessage,${PHONES1VM})
exten = s,3,Hangup
Hope this helps.
Kind Regards
Stuart
On Monday, Jan 24, 2005, at 21:26 Australia/Perth, Remco Barende wrote:
Cool, is this equal if you use ringer groups or queues?
At home it's a bit silly to create a call queue that is answering and
put a caller in queue. I just want several phones to ring and only
answer the phone when somebody is picking it up.
I tried just specifying several sip devices to ring but if one of all
the sip phones specified is not registered any call will immediately
be forwarded to voicemail.
Thanks!
On Mon, 24 Jan 2005, Jon Radon wrote:
This is easy as pie. Just Dial without an Answer. When the device
you specified in dial picks up, it will pick up the line. Done and
done.
On Mon, 24 Jan 2005 20:40:04 +0800, Stuart Elvish
[EMAIL PROTECTED] wrote:
Hi,
I was just wondering whether or not anybody has a dial plan or some
advice on getting a SIP phone to ring without answering the PSTN line
so that the caller doesn't have to pay for the phone call unless it
actually get answered by a human or the answering machine after 40
seconds. I had a look through the wiki but there wasn't anything I
could find (probably the wrong search terms). Any advice is greatly
appreciated.
Kind Regards
Stuart
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Re: [Asterisk-Users] Auto callout - reminder - is it possible?
On Mon, 24 Jan 2005, Andrew Kohlsmith wrote: As far as integrating with a website or database -- that is a piece of cake. Your backend logic just determines when a call is needed and gerates the approprate .call file. Just remember to create it in /tmp or something, close it and then MOVE it to the outgoing spool instead of creating and working on it in the outgoing spool. You need to create the temporary file on the same device as the call spool resides on. Otherwise the move from the temporary location the the call spool will not be an atomic operation but rather a read-write-unlink sequence. This has been discussed earlier on the mailing list. Just make a temporary directory next to the call spool directory and create the files there. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] forwarding sip
I'm using asterisk to forward some sip incoming calls to ser, I've noticed that every call * passes to ser has sip:[EMAIL PROTECTED] as header. Is there a way to make * pass the number of the caller in sip address to ser. I mean if I get a call from PSTN number 123456, how can I pass sip:[EMAIL PROTECTED] to ser? In docs I found fromuser, username, callerid options, but they affect only the from part preceeding real sip address hope anyone can help me thanks a lot ciao -- Giovanni Balasso [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Any experience with Sangoma cards?
Robert - The Sangoma cards work well with *. Look for some announcements from Sangoma in the coming months they have a few things they are working on which will help them compete with Digium in the * market. Cory Andrews Senior Partner VOIPSupply.com + 800.398.VOIP X22 [EMAIL PROTECTED] Robert Augustyn wrote: Hi, I am considering A101/102/104 cards for my asterisk installations. Has any of you used these or any Sangoma cards in such environment? Any thoughts? How do they stack up against Digium cards? Any input would be greatly appreciated. robert ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAXTEL is dead/dying?
As someone that's just recently setup an * server I agree. I thought about setting up an Iaxtel account as well but couldn't see the point in it because I had setup FWD for testing. I continue to use FWD for all my toll free calls and the quality is just awesome. I can't see how Iaxtel would provide any additional benefit. Perhaps the time for Iaxtel has come and gone. There are plenty of IAX2 providers these days, * has become quite popular, so the need for a separate telecom network doesn't make a whole lot of sense; not that FWD isn't separate, it's just more popular IMHO. -mark On Jan 21, 2005, at 6:12 PM, Michael Graves wrote: Yeah, FWD has been pretty good about their beta of the IAX2 support. My * server has been on it for 6 months without too much trouble. I even use it to bridge out to Signate.co.uk where my boss has an account. It was crystal clear last night from Houston TX to Cambridge UK. Dead reliable. I'm dropping my IAXTel registration when next I get around to such things. -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk with Grandstream ringback
Hi All We have Grandstream 102's running ver X.18. When hanging up after a call has been made the grandstream seems not to disconnect the call and when you put the handset down the phone rings only to pick it up and be on the same call. This is happening quite often and gets very irritating. Can anyone help with this? Regards Doug ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_skinny and firmware upgrade
Monday, January 24, 2005, 9:23:50 AM, Steve Hanselman wrote: Nothing to do with skinny, drop the new file(s) in your tftp directory and edit the .xml file to specify the new version, the phone will upgrade itself when it loads the config. the firmware I have doesn't request xml file it requests SEPMAC.cnf I udnerstand the new versions of firmware request SEPMAC.cnf.xml. Not sure where to go from here, any ideas? -- Best regards, Subhi S Hashwamailto:[EMAIL PROTECTED] When everything is heading your way, you're in the wrong lane. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TDM400P Sync source
Hi, I am trying to track down the reason to my problems with sending and reciving fax with my PRI and 2 TDM400P Cards: PSTN - PRI (E100P) - * - TDM400P - Fax Machine I have used Zapbarge to listen to the data stream, but I can't say if it really have some time slips - fax kinda noisy in itself. Using the zttool i saw the Sync source for the TDM are internally - what does that mean? Are they using an on-board source, or using the PRI (which is configured to use the telco as sync source). The zttool reports this for the E100P: Sync Source:Digium Wildcard E100P E1/PRA C For the TDM400P cards it reports : Sync Source:Internally clocked Is there a way to specify which source the TDM cards should use? Div system info: System: SuperMicro with P4 2.53GHz 512 MB Ram 3Ware IDE Raid Cont. (Running Raid 5) 1 x E100P 2 x TDM400 Kernel-2.6.8.1 (No ACPI) Asterisk, Libpri and zaptel is from stable 1.0.2 release zaptel.conf span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 fxoks=32,33,36-39 fxsks=34,35 cat /proc/interrupts CPU0 0: 954710104 XT-PIC timer 1: 20 XT-PIC i8042 2: 0 XT-PIC cascade 5: 954609651 XT-PIC wctdm 7: 954739525 XT-PIC wctdm 10:2858318 XT-PIC eth1 11: 32640181 XT-PIC eth0 12:214 XT-PIC i8042 14: 954602423 XT-PIC t1xxp 15:5550321 XT-PIC 3ware Storage Controller NMI: 0 LOC: 954781513 ERR: 0 MIS: 0 -- Med venlig hilsen / Best regards Michael Løjtnant - Systems Engineer ZyXEL Communications A/S Columbusvej 5 - 2860 Søborg Tel (+45) 3955 0700 - Fax (+45) 3955 0707 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX Inbound Sound Quality
Try changing to a less-bandwidth intensive codec (like GSM) and see what happens. -mark On Jan 21, 2005, at 7:08 PM, Brian Dingman wrote: I have a couple of DID's through VP Connect and have been having sound quality issues on incoming calls. During the call, the calling parties voice sometimes sound like it is crackling, in other words it is not very crisp. I would liken it to listening to a radio with a blown speaker. This sound defect comes and goes throughout the call. The other person is always audible but it just isn't as crisp and clear as when I make outgoing calls over IAX. The other party does not hear any audio defects. Anybody have any suggestions on tweaking this? Or has anyone experienced the like? Running * 1.0.3 on an AMD 1700 with 512 MB of RAM (Red Hat 9). I am the only user currently on the system. I am connecting with their IAX server using ULAW and my SIP phone is also using ULAW (Sipura 2000). Thanks, Brian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Auto callout - reminder - is it possible?
On January 24, 2005 09:36 am, Peter Svensson wrote: You need to create the temporary file on the same device as the call spool resides on. Otherwise the move from the temporary location the the call spool will not be an atomic operation but rather a read-write-unlink sequence. This has been discussed earlier on the mailing list. This is true. On my systems /var/spool is on the same drive as /tmp, except in the case where /tmp is a ramdisk but I typically don't do that, as RAM is better used for memory. :-) -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco7905 keeps forwarding to voicemail
Hello All! I have a strange problem with Cisco 7905. It is forwarding unanswered calls to VoiceMail even thought I have setup it not to. My ring timer on cisco 7905 is 60s, and my ForwardToVMDelay is 3000s. This means that call should never be forwarded to VM! This is true if I call from internal number then this happens on asterisk: -- SIP/104-6073 is ringing -- Nobody picked up in 6 ms -- Executing Busy(SIP/100-865d, ) in new stack == Spawn extension (normal, 104, 2) exited non-zero on 'SIP/100-865d' -- Executing Hangup(SIP/100-865d, ) in new stack == Spawn extension (normal, h, 1) exited non-zero on 'SIP/100-865d' But if I call from External ISDN line this happens: -- SIP/104-19cc is ringing -- Got SIP response 302 Moved Temporarily back from 192.168.10.154 -- Now forwarding CAPI[contr3/2347474]/23 to 'Local/[EMAIL PROTECTED]' (thanks to SIP/104-19cc) -- Executing Answer(Local/[EMAIL PROTECTED],2, ) in new stack -- Executing Wait(Local/[EMAIL PROTECTED],2, 1) in new stack -- Local/[EMAIL PROTECTED],1 answered CAPI[contr3/2347474]/23 -- CAPI Answering for MSN 2347474 == Spawn extension (limited, 104, 1) exited non-zero on 'CAPI[contr3/2347474]/23MASQ' -- Executing Hangup(CAPI[contr3/2347474]/23MASQ, ) in new stack == Spawn extension (limited, h, 1) exited non-zero on 'CAPI[contr3/2347474]/23MASQ' -- Executing VoiceMailMain(CAPI[contr3/2347474]/23, s040684543) in new stack -- Playing 'vm-login' (language 'en') As I understand this Cisco is saying back to Asterisk 302 Moved Temporarily and forwards call to 850. This should happen because it configured not to forward! Any ideas? Br, Alen ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: IAX Inbound Sound Quality
On Jan 22, 2005, at 10:49 PM, Michael Graves wrote: I notice that all four of my IAX2 based termination providers send incomming calls in trunking mode. You can tells since the command IAX2 Show Registry reports all the connections to port 8617. This is something that is determined at their end. In trunk mode I beleive that the jitter buffer is not effective. IIRC the jitter buffer is currently broken in trunk mode and should be turned off. http://www.voip-info.org/wiki-Asterisk+bandwidth+iax2 An alternative for testing is to set trunk=no in iax.conf. I've had to do that for my VPC trunks because I've also found that outbound faxing seems to be broken with trunking turned on (at least to VPC). FWIW, I had similar problems with VPC so I switched to Sixtel.net. No such problems anymore. VPC must still be using quite a lot of custom code or routing their calls in some weird way because I've found two problems with them so far while using IAX2: 1) The fax problem mentioned above. 2) Inbound DTMF is quite broken. (They are working on a fix and said it would be at least 30 days...but then in December they said it would take 2 weeks...). What a drag. -mark -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] how to use mysql with asterisk
hello i want to use mysql database server with my asterisk PBX. i have installed mysql on linux mechine. i have already installed asterisk on same mechine. now i want to know what is the way to connect asterisk to mysql. __ Do you Yahoo!? Take Yahoo! Mail with you! Get it on your mobile phone. http://mobile.yahoo.com/maildemo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Any experience with Sangoma cards?
[EMAIL PROTECTED] wrote: Hi, I am considering A101/102/104 cards for my asterisk installations. Has any of you used these or any Sangoma cards in such environment? Any thoughts? How do they stack up against Digium cards? Any input would be greatly appreciated. robert The Sangoma cards are very well built and very well supported. They've been in business since 1984, and are commited to Linux and open source. The T1 cards they have for Asterisk are ASIC-based, which means that they can program their card to be whatever they want it to be. For Zaptel compatibility, they actually dumbed-down their cards - the hardware is capable of a lot more. Expect to see exciting things from Sangoma with respect to open source telephony. Cheers, Jim. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.7.2 - Release Date: 21/01/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Short DTMF Tones and Asterisk
Have you tried calling into * using another phone or phone system? Try it from a cell. Try it from the pub, etc. etc. It may have nothing to do with the length of the DTMF at all because IAX2 sends DTMF out of band. FWIW, inbound DTMF is not working properly with Voicepulse Connect either right now when using IAX2 (digits are missed), but everything seems to work fine over FWD via IAX2 (at least the last time I checked). -mark On Jan 24, 2005, at 5:31 AM, Robert P. McKenzie wrote: I'm having a very annoying problem with access my asterisk system from work. Our phone system here only produces very very short DTMF tones. The phones work fine for other IVR systems (Dell Support, HP Support, etc, etc). However, tones to Asterisk just never make it. The way I'm calling into my Asterisk server is such: OFFICE PHONE = CALLUK.COM 0870 = IAX Inbound The phone quality of the spoken call is fine, but DTFM tones aren't working. I'm using ulaw as the codec and bandwidth has been set to high in iax.conf. Any advice would be great. I could post debug logs of a call if someone would care to explain exactly what to capture. I'm still a newbie to Asterisk. Thanks in advance. -- Robert P. McKenzie | GammaRay Technical Services Ltd [EMAIL PROTECTED] | [EMAIL PROTECTED] http://www.uk-experience.com | http://www.gammaray-tech.com Ecademy Profile: http://www.ecademy.com/account.php?op=viewid=64014 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Sipura Behind NAT howto
I am trying to get a SPA-3000 to work behind NAT - for the sake of the exercice. Post the relevant entries from sip.conf and extensions.conf, and the relevant fields from the SPA-3000 Line 1 tab. -- Nabeel Jafferali Tel: +1 (416) 628-9342 Toronto +1 (646) 225-7426 New York FWD: 46990 Email/MSN: nabeelatjafferali.net ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IVR Timing out
I set up an IVR systems that plays a message for 15 seconds but once the message is over you can not select any of the prompts. If you select something within 10 seconds the IVR system works. I even set the ResponseTimeout cmd to 25 secs but that does not work. Jan 24 09:54:29 NOTICE[-1222644816]: sched.c:221 sched_settime: Request to schedule in the past?!?! [attendant] ;Main welcome message exten = s,1,Wait(2) exten = s,2,DigitTimeout,5 exten = s,3,ResponseTimeout,25 exten = s,4,Background(welcome_n2p1) exten = s,5,Hangup Thanks in advance for help, Kurt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] XEON or not
Are there much performance differences when using XEON or not? In my case, I will go with muLaw both in and out of Asterisk. Are there really any processing at all if it's using same codec all the way? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] zaptel vanilla kernel
hi, to digium maybe some individuals: do you plan add zaptel drivers to vanilla kernel? for users is this very good thing --- Marek Cervenka Centrum Vypocetni Techniky CVT - http://cvt.fpf.slu.cz FPF SLU OPAVA - http://www.fpf.slu.cz LCNA- http://lcna.slu.cz === ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Threeway callin
Do you mean conferencing? or local conferencing? Local is usualy implemented by your phone. and conferencing is done in * using meetme command. Check the wiki On Mon, 24 Jan 2005 18:57:03 +0530, Ritesh Jalan [EMAIL PROTECTED] wrote: Can eny body tell me how to configure threeway calling using SIP channels? Thanks Regards Ritesh ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] how to display queue status and/or line status in asterisk
What is the best way to display queue status to a station? Our current phone system has 4 lines at each station so it is really easy to see how many lines are waiting. I could replicate this using a 4 line phone, but this requires both running an extra 4pair to each desk as well as taking up 4 slots on a channel bank for each extension, so this is not a very feasible solution. Is there a way to maybe have virtual lines on an ADSI phone or using callerid to somehow relay the number of people waiting in a queue? How are other people dealing with line/queue status in asterisk? Thanks, Jon. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zapata in Australia
On Mon, 2005-01-24 at 21:37, Emanuele Venditti wrote: Does anybody what the regional settings are to use an x100p (clone) card with Asterisk in Australia? I got mine installed and recognised by * but I get no sound and terrible hangup detection. Basically after each test call to the landine number (plugged into the x100p card) I need to unplug the cord and plug it back in to get a normal dialtone. When * answers the call (or diverts it to any internal IP phone) there is absolutely no sound. This works for me in AU. In /etc/zaptel.conf: fxsks=1 loadzone = au defaultzone=au In /etc/asterisk/zapata.conf: [channels] context = default signalling = fxs_ks echocancel = 128 echocancelwhenbridged = yes echotraining = yes relaxdtmf = yes pulsedial = yes rxgain = +15% txgain = +5% immediate = no busydetect = yes busycount = 3 callprogress = yes musiconhold = default usecallerid = yes callerid = asreceived useincomingcalleridonzaptransfer = yes faxdetect = both group = 1 channel = 1 Note that I do not get callerid but I do get fax. many thanks manny __ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ISP connection to the PSTN using Asterisk
Hi all, Could someone let me know the most common way that an Internet ISP would allow customers access to the PSTN?? Do they buy multiple fxo cards such as the TDM400P and rent multiple lines from a larger provider?? Would the best way be to connect to a third party voice/pstn gateway?? Is that simply a matter of forwarding all sip traffic destined for the pstn to another provider with a gateway and then they have to worry about the number of lines etc??And if that is the case, I presume no extra hardware is required? Thanks, Aisling. ---Legal Disclaimer--- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] how to use mysql with asterisk
Hi p_kami, You need cdr_addon_mysql.so on your /usr/lib/asterisk/modules/ and to config the cdr_mysql.conf file configured. Here is an example: [global] hostname=localhost dbname=yourserverip user=mysqluser password=userpwd ;port= ;sock= of course you will need MySQL and a table for the cdr. Mensaje citado por Kamran Ahmad [EMAIL PROTECTED]: hello i want to use mysql database server with my asterisk PBX. i have installed mysql on linux mechine. i have already installed asterisk on same mechine. now i want to know what is the way to connect asterisk to mysql. __ Do you Yahoo!? Take Yahoo! Mail with you! Get it on your mobile phone. http://mobile.yahoo.com/maildemo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Alfred Certain Yance GECKO Visite: www.gecko-soft.com Av 15 No 106 50 Of 403 Bogota D.C. - Colombia Tel: +1 6127092 This e-mail and attachments, if any, may contain confidential and/or proprietary information. Please be advised that the unauthorized use or disclosure of the information is strictly prohibited. If you are not the intended recipient, please notify the sender immediately by reply e-mail and delete all copies of this message and attachments. Thank you. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] XEON or not
Hello, Your on the right track. No translations = no (well, not much) CPU load. If all your doing is pushing packets, the NIC will become more important than the CPU. Don't forget, MOH, auto response and VM will take some load as these may need to be transcoded, unless you save them in the same codec (format) as the native voice traffic. For under 20 concurrent calls, a P4 1Ghz would do. Jon - Original Message - From: Daniel Nyström [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Monday, January 24, 2005 10:19 AM Subject: [Asterisk-Users] XEON or not Are there much performance differences when using XEON or not? In my case, I will go with muLaw both in and out of Asterisk. Are there really any processing at all if it's using same codec all the way? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Any experience with Sangoma cards?
Jon, Would you care to comment on how have you been using these? Thanks robert -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jon Bebeau Sent: Monday, January 24, 2005 8:52 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Any experience with Sangoma cards? Hello, Yes. I've had good experience with all three you mentioned. Jon - Original Message - From: Robert Augustyn [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Sunday, January 23, 2005 2:17 PM Subject: [Asterisk-Users] Any experience with Sangoma cards? Hi, I am considering A101/102/104 cards for my asterisk installations. Has any of you used these or any Sangoma cards in such environment? Any thoughts? How do they stack up against Digium cards? Any input would be greatly appreciated. robert ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] XEON or not
Hi Daniel, I got better performance using Xeon for QuadPRI. Alfred Certain Yance GECKO Visit: www.gecko-soft.com Av 15 No 106 50 Of 403 Bogota D.C. - Colombia Tel: +1 6127092 This e-mail and attachments, if any, may contain confidential and/or proprietary information. Please be advised that the unauthorized use or disclosure of the information is strictly prohibited. If you are not the intended recipient, please notify the sender immediately by reply e-mail and delete all copies of this message and attachments. Thank you. Are there much performance differences when using XEON or not? In my case, I will go with muLaw both in and out of Asterisk. Are there really any processing at all if it's using same codec all the way? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dialing Delay
Hello, When I dial out there is a long delay in dialing. Is this normal? Thanks, David ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] zaphfc no callerid incoming to SIP phone butvisible in logfile
Check this out.
http://lists.digium.com/pipermail/asterisk-users/2005-January/084942.html
On Mon, 24 Jan 2005 13:14:33 +0100, Rob Scott [EMAIL PROTECTED] wrote:
Try commenting out the line
pritrustusercid = yes
Or set it to 'no'.
That worked for me.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jens
Sent: Friday, January 21, 2005 7:18 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] zaphfc no callerid incoming to SIP phone
butvisible in logfile
Hi,
I think you didn't set usecallerid=yes in your zapata.conf?
Another way is to set the callerid in your extensions.conf via exten =
807440,2,SetCIDNum(0${CALLERIDNUM}). So you also have a 0 in front of the
displayed number - nice for callback.
regards
Jens
Hello,
I've added a ZAPHFC card to my CAPI based system. Calls coming in via
ZAPHFC do not forward the caller id to the SIP phones. Calls coming in
via CAPI do forward the caller id to the SIP phones.
--
Jens Lentföhr
http://www.jens-it.de
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RE: [Asterisk-Users] PSTN and Asterisk
Title: Message www.voip-info.org www.asteriskdocs.org -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Vassili GontcharovSent: January 24, 2005 5:13 AMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] PSTN and Asterisk Hi quys, I look for a solution for interconnection beetwen PSTN and VoIP. My application have to treat few protocols comming from PSTN lines and mixing data , dtmf and voice. Can I use Asterisk for : PSTN-- Asterisk (converting analog call to IP) -- MyApplication( translation protocolsand do some workswith incomming data) What hardware I can use for this? Do use Asterisk G.711 protocol? Thanks Vassili --No virus found in this incoming message.Checked by AVG Anti-Virus.Version: 7.0.300 / Virus Database: 265.7.2 - Release Date: 21/01/2005 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.7.2 - Release Date: 21/01/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] chan_skinny and firmware upgrade
From the very early days of Cisco skinny the phones have all requested XMLDefault.cnf.xml, you just need to pop it in there (either run a tcpdump on the tftp port or run the daemon in logging mode and you'll see). Steve -Original Message- From: Subhi S Hashwa [mailto:[EMAIL PROTECTED] Sent: 24 January 2005 14:46 To: Steve Hanselman Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [Asterisk-Users] chan_skinny and firmware upgrade Monday, January 24, 2005, 9:23:50 AM, Steve Hanselman wrote: Nothing to do with skinny, drop the new file(s) in your tftp directory and edit the .xml file to specify the new version, the phone will upgrade itself when it loads the config. the firmware I have doesn't request xml file it requests SEPMAC.cnf I udnerstand the new versions of firmware request SEPMAC.cnf.xml. Not sure where to go from here, any ideas? -- Best regards, Subhi S Hashwamailto:[EMAIL PROTECTED] When everything is heading your way, you're in the wrong lane. The information contained in this email is intended for the personal and confidential use of the addressee only. It may also be privileged information. If you are not the intended recipient then you are hereby notified that you have received this document in error and that any review, distribution or copying of this document is strictly prohibited. If you have received this communication in error, please notify Brendata immediately on: +44 (0)1268 466100, or email '[EMAIL PROTECTED]' Brendata (UK) Ltd Nevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX UK Registered Office as above. Registered in England No. 2764339 See our current vacancies at www.brendata.co.uk ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] UPS for Asterisk
-Original Message- From: Shoval Tomer [mailto:[EMAIL PROTECTED] On the other hand, telephony down time is unacceptable. PBXs have a counter part. Plain old PBXs are expected to run 24x7. real 24x7, with uptimes of 99.999. And if you think about it, they actually do. That would be news to the people who installed our (non-Asterisk) PBX. It has no battery backup at all. When the power goes out, so do all our phones. (Except for the fax machines, which don't go through the PBX.) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_skinny and firmware upgrade
Monday, January 24, 2005, 3:38:57 PM, Steve Hanselman wrote: From the very early days of Cisco skinny the phones have all requested XMLDefault.cnf.xml, you just need to pop it in there (either run a tcpdump on the tftp port or run the daemon in logging mode and you'll see). I did, that is how i foundout about the SEPMAC.cnf and SEPDefaults.cnf Ethernet II, Src: 00:0a:8a:f9:09:10, Dst: 00:50:8b:75:4b:fb Destination: 00:50:8b:75:4b:fb (CompaqCo_75:4b:fb) Source: 00:0a:8a:f9:09:10 (Cisco_f9:09:10) ... Trivial File Transfer Protocol Opcode: Read Request (1) Source File: SEP000A8AF90910.cnf Type: octet [Malformed Packet: TFTP] The xml request is a feature of new firmware, that is my guess. -- Best regards, Subhi S Hashwamailto:[EMAIL PROTECTED] When everything is heading your way, you're in the wrong lane. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] budgetone - pattern matching for ringtones - firmware 1.0.5.18
Hi, It seems the patter matching on CallerID rule is an exact matching with this firmware. ie: if you configured "30" for 2nd ringtone then callerID "30" will match and callerid "301" will NOT match. This doesn't correspond to the wiki description ( http://www.voip-info.org/wiki-Asterisk+phone+grandstream+budgetone). Can anybody confirm? Is there a workaround (the older bahaviour was more flexible)? Tx, Luka. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] UPS for Asterisk
-Original Message- From: Jon Radon [mailto:[EMAIL PROTECTED] Why risk it? Just go snag a cheap UPS from your local store. Just get something with enough run time to shut the system down gracefully. Don't go *too* cheap, though. I had a couple of really cheap (under $40) CyberPower UPS's that ended up causing more outages than they protected against. I've had good luck with APC, but keep in mind that the batteries have a finite lifespan. On SmartUPS and BackUPS Pro models, you'll get a warning that the battery needs replacing, but on regular BackUPS models the first hint you get that the battery is bad is when the power goes out and the UPS doesn't work. This is sometimes okay for workstation use, but I'd hesitate to put one of those on a server. I find that the batteries in our APC UPS's generally last four to five years for stand-alone units, three years for rack-mount ones. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Best VPN server for * and woad warriors usin g windows?
-Original Message- From: Remco Barende [mailto:[EMAIL PROTECTED] Why is it bad to put a vpn server on the * box? CPU load. IPSec can be quite CPU intensive. So can asterisk. Putting two CPU-intensive, time-sensitive applications on one machine is asking for trouble. It may work, though, if you don't have too many simultaneous users. I was considering IPSEC because I heard it is safer or more secure than PPTP and Windows XP supports IPSEC natively (or so it claims). It does, but I've never had much luck getting it to interoperate with anything but a Windows server. I've heard it can be done, but I don't understand IPSec well enough to make it work. It's not simple. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Mobile Callings
Hi, Does someone knows what kind of device I need to call from my pc to the mobile network? In Spain VoIP prices are very similar to call to a mobile than do it from an other mobile. So, I want to plug some device to the PC and get out the call throught it, but I dn't know what kind of device I need. Thanks in advance ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] UPS for Asterisk
Why would the heads come in contact with the platters on a powerfail? The arms are very rigid -- the heads only float a few thousandths of an inch over the platters -- something that I don't believe has anything to do with the platters spinning (that may *help* but I don't think the heads will contact Search google on Bernoulli Effect ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] can the dialtone be changed after pressing 9?
Alexander (or anyone),
Can you point me to where this can be done for zap
devices? zonedata.c, perhaps? How?
Thanks,
Greg
Yes you can but it only works for zap devices. IP based would be a
function of the hardware. extensions.conf
has ignorepat = 9 exten =
_9X.,1,Dial(Zap/G2/${EXTEN:1}) The first user to try it
asked if instead of keeping the same dialtone after pressing 9, if
I could play a different dialtone. Can this be done?
I'm running asterisk 1.0.0 in case that matters.___
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Re: [Asterisk-Users] Zapata in Australia
As a general rule, the X100P should not be used in Australia as it is set to an incorrect impedence and can't be changed. The TDM series of cards with FXO/FXS modules can be set to work in AU. ... You should also be aware that the PSTN connect cards do not have Austel approval as yet, and so they shouldn't be connected the the public phone network. Andrew On 25/01/2005, at 2:25 AM, Howard Lowndes wrote: On Mon, 2005-01-24 at 21:37, Emanuele Venditti wrote: Does anybody what the regional settings are to use an x100p (clone) card with Asterisk in Australia? I got mine installed and recognised by * but I get no sound and terrible hangup detection. Basically after each test call to the landine number (plugged into the x100p card) I need to unplug the cord and plug it back in to get a normal dialtone. When * answers the call (or diverts it to any internal IP phone) there is absolutely no sound. This works for me in AU. In /etc/zaptel.conf: fxsks=1 loadzone = au defaultzone=au In /etc/asterisk/zapata.conf: [channels] context = default signalling = fxs_ks echocancel = 128 echocancelwhenbridged = yes echotraining = yes relaxdtmf = yes pulsedial = yes rxgain = +15% txgain = +5% immediate = no busydetect = yes busycount = 3 callprogress = yes musiconhold = default usecallerid = yes callerid = asreceived useincomingcalleridonzaptransfer = yes faxdetect = both group = 1 channel = 1 Note that I do not get callerid but I do get fax. many thanks manny __ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Menu tree for voicemailmain application
Is there a menu tree diagram somewhere for the Voicemailmain application? I know my users will ask for one, and before I started drawing my own I thought I'd see if someone already had. --- David Brodbeck, System Administrator InterClean Equipment, Inc. 3939 Bestech Drive Suite B Ypsilanti, MI 48197 (734) 975-2967 x221 (734) 975-1646 (fax) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] UPS for Asterisk
On Monday 24 January 2005 02:52, Peter Svensson wrote: On Sun, 23 Jan 2005, Andrew Kohlsmith wrote: Why would the heads come in contact with the platters on a powerfail? The arms are very rigid -- the heads only float a few thousandths of an inch Well, I'm sorry but I find this whole discussion on why you should have a UPS a bit silly. Electronics are sensitive to ... electricity. May it come in sudden drops just as the data is only in cache someplace, or pulsing power going on and off and back on. Never mind spikes. Fortunately we have pretty good equipment these days that can handle a lot of abuse. But why would anyone argue against it? Either you have the money for it or not. The chance of loosing equipment is there either way. Buy a good UPS and use it if you can. Period. The days of shoddy UPS's are long gone, unless you always buy the cheapest stuff you can find all the time. In which case you might be able to find something crappy. APC gives good support and make decent UPS's at a decent price. -- Steve Szmidt They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dialing Delay
On Mon, 2005-01-24 at 07:42 -0800, David Shaw wrote: Hello, When I dial out there is a long delay in dialing. Is this normal? No it isn't normal. Examine/post relevant portions of config files and explain what interfaces you are using. Quick guess is the pattern match for your outbound calls is waiting for a timeout instead of matching a real specific pattern. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 1.0.4 and more ... + rpm spec
Will be good, if somebody could provide rpms for every release and also rpm's with static compiled chan_oh323 and Asterisk-oh323 modules asterisk.spec for 1.0.5 is in attachment put this file into /usr/src/redhat/SPECS asterisk-1.0.5.tar.gz to the /usr/src/redhat/SOURCES cd /usr/src/redhat/SPECS rpmbuild -ba asterisk.spec if this file will be contained directly in the tarball (like openvpn or other good software), then simply run rpmbuild -ta asterisk-1.0.5.tar.gz --- Marek Cervenka Centrum Vypocetni Techniky CVT - http://cvt.fpf.slu.cz FPF SLU OPAVA - http://www.fpf.slu.cz LCNA- http://lcna.slu.cz === ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] PRI dchannel in use? (take 2)
I just started getting this error today (I've gotten this error befor) and its preventing me from having any incoming calls: chan_zap.c:7542 pri_dchannel: Ring requested on channel 0/2 already in use on span 1. Hanging up owner. PRI has been working fine. I didn't know anything was wrong until someone came and said their DID wasn't working. You call their DID, asterisk shows the message above and you get fast busy. But I can make outgoing calls no prob. I'm running the latest stable versions of libpri and zaptel and asterisk. any ideas? Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
