[Asterisk-Users] Strange softphone issue - audio open before answer
Hi All... I'm not sure if this is a bug or a feature. When I use a soft phone such as iaxcomm and firefly, I find that when the extension is rung from any channel (zap, IAX, SIP) that while the phone is ringing, before it is answered, audio is passed between the caller and called phone. This even happens when people call from the outside world. Is there a way to stop this? Thanks very much... ___ 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] MeetMe problem - some parameters ignored
Hi All... I set up a conference bridge using MeetMe. It works nicely, except that it seems that certain parameters I give it are ignored or else don't work. Here is the line from my dial plan: exten = 6500,1,absolutetimeout,0 exten = 6500,2,MeetMe,100|ciMpPs|1234 The MOH and * work, but users are not announced when they join or leave and the pin is not requested. Maybe I am misunderstanding what these are supposed to do? ___ 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] Time out not working from php agi...
Here i am doing a dial command from a php agi... EXEC DIAL H323/[EMAIL PROTECTED]:1720|40|HL(585000:61000:3) But asterisk is not disconnecting the connection after 585 secs... the result is ... answered time is 1926n but thing is time out is working some time and some time not LOG: 2005-06-28 20:26:13 VERBOSE[19094] logger.c: callcard.php: string(111) app_callingcard: Dialing 'H323/[EMAIL PROTECTED]:1720| 2005-06-28 20:26:13 VERBOSE[19094] logger.c: callcard.php: EXEC DIAL H323/[EMAIL PROTECTED]:1720|40|HL(585000:61000:3) 2005-06-28 20:26:13 VERBOSE[19094] logger.c: AGI Script Executing Application: (DIAL) Options: (H323/[EMAIL PROTECTED]:1720|40|HL(585000:61000:3)) 2005-06-28 20:26:13 VERBOSE[19094] logger.c: -- Limit Data: 2005-06-28 20:26:13 VERBOSE[19094] logger.c: -- timelimit=585000 2005-06-28 20:26:13 VERBOSE[19094] logger.c: -- play_warning=61000 2005-06-28 20:26:13 VERBOSE[19094] logger.c: -- play_to_caller=yes 2005-06-28 20:26:13 VERBOSE[19094] logger.c: -- play_to_callee=no 2005-06-28 20:26:13 VERBOSE[19094] logger.c: -- warning_freq=3 2005-06-28 20:26:13 VERBOSE[19094] logger.c: -- start_sound=UNDEF 2005-06-28 20:26:13 VERBOSE[19094] logger.c: -- warning_sound=timeleft 2005-06-28 20:26:13 VERBOSE[19094] logger.c: -- end_sound=UNDEF 2005-06-28 20:26:13 DEBUG[19094] chan_h323.c: type=H323, format=4, [EMAIL PROTECTED]:1720. 2005-06-28 20:26:13 DEBUG[19094] chan_h323.c: Extension: 880178034593 Host: xx.xx.xx.xx:1720 2005-06-28 20:26:13 DEBUG[19094] chan_h323.c: Could not find peer xx.xx.xx.xx by name or address 2005-06-28 20:26:13 DEBUG[19094] chan_h323.c: Placing outgoing call to [EMAIL PROTECTED]:1720, 101 2005-06-28 20:26:13 VERBOSE[19094] logger.c: -- Called [EMAIL PROTECTED]:1720 2005-06-28 20:58:41 DEBUG[19094] channel.c: Didn't get a frame from channel: IAX2/[EMAIL PROTECTED]:4569-13 2005-06-28 20:58:41 DEBUG[19094] channel.c: Bridge stops bridging channels IAX2/[EMAIL PROTECTED]:4569-13 and H323/ 2005-06-28 20:58:41 DEBUG[19094] app_dial.c: Exiting with DIALSTATUS=ANSWER. 2005-06-28 20:58:41 VERBOSE[19094] logger.c: callcard.php: GET VARIABLE ANSWEREDTIME 2005-06-28 20:58:41 VERBOSE[19094] logger.c: callcard.php: GET VARIABLE DIALSTATUS 2005-06-28 20:58:41 VERBOSE[19094] logger.c: callcard.php: GET VARIABLE DIALEDTIME 2005-06-28 20:58:41 VERBOSE[19094] logger.c: callcard.php: string(31) res is , answered time is 1926nn 2005-06-28 20:58:41 VERBOSE[19094] logger.c: callcard.php: string(33) res is , dialedtime time is 1948nn 2005-06-28 20:58:41 VERBOSE[19094] logger.c: callcard.php: string(15) Stop time: 0506n 2005-06-28 20:58:41 VERBOSE[19094] logger.c: callcard.php: string(36) Disconnect time: 2005-06-28 20:58:41n 2005-06-28 20:58:41 VERBOSE[19094] logger.c: callcard.php: CHANNEL STATUS IAX2/[EMAIL PROTECTED]:4569-13 2005-06-28 20:58:41 VERBOSE[19094] logger.c: callcard.php: result is 6 2005-06-28 20:58:41 WARNING[19094] file.c: Failed to write frame 2005-06-28 20:58:41 VERBOSE[19094] logger.c: == Spawn extension (default, 1112, 3) exited non-zero on 'IAX2/[EMAIL PROTECTED]:4569-13' 2005-06-28 20:58:41 DEBUG[19094] chan_iax2.c: We're hanging up IAX2/[EMAIL PROTECTED]:4569-13 now... 2005-06-28 20:58:41 DEBUG[19094] chan_iax2.c: Really destroying IAX2/[EMAIL PROTECTED]:4569-13 now... 2005-06-28 20:58:41 VERBOSE[19094] logger.c: -- Hungup 'IAX2/[EMAIL PROTECTED]:4569-13' ___ 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 + spandsp
On Sat, Jul 09, 2005 at 11:43:00PM +, Leonardo F. Bauchwitz wrote: Hello Tzafrir: Tzafrir Cohen wrote: On Sat, Jul 09, 2005 at 01:15:16PM +, Leonardo F. Bauchwitz wrote: Hello: I dont know, if is my question to do hier, or in the dev-list, but anyway: I 've installed Asterisk (head, development because I need Realtime), but when I try to apply the patch I 've got many errors, reason why I wrote myself the apps/Makefile. (Of course, first, I compiled spandsp, etc.) Then, I try to compile Asterisk, but it 's impossible: For the record, the debian source package asterisk-apps-spandsp builds out-of-tree just fine. I use Debian and Ututo-e (and I have proved Xorcom :)), but this package, -asterisk-apps-spandsp- support Asterisk Real Time? The package itself is a simple asterisk application. I don't think that real-time configuration is much relevant to it. Maybe you need to build it with HEAD to get real-time, though. Now, I work with the development version of Asterisk because support that issue. We don't yet package HEAD, however the above was a general comment about out-of-tree building of asterisk modules. The amount of patching needed to get them to build out-of-tree is generally minimal. There's a small script in the contrib directory to help with that, but so far I never used it. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ 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: MeetMe problem - some parameters ignored
In article [EMAIL PROTECTED], Jim Archer [EMAIL PROTECTED] wrote: Hi All... I set up a conference bridge using MeetMe. It works nicely, except that it seems that certain parameters I give it are ignored or else don't work. Here is the line from my dial plan: exten = 6500,1,absolutetimeout,0 exten = 6500,2,MeetMe,100|ciMpPs|1234 The MOH and * work, but users are not announced when they join or leave and the pin is not requested. Maybe I am misunderstanding what these are supposed to do? You need to read about the difference between CVS-HEAD (development version) and CVS-STABLE (the 1.0.x series). Some of the above options (including 'i'), and also the 'r' option in your other posting, only exist in the development version, not in the 1.0.x versions, as they were added after the 1.0 feature freeze was made. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ 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] IAXphone - ip address - extension number.
Yeee-h! doesn't this look pretty? *** Asterisk Ready. *CLI -- Accepting AUTHENTICATED call from 192.168.0.201, requested format = 4, actual format = 4 -- Executing Dial(IAX2/[EMAIL PROTECTED]/1, IAX2/z2|20|tr) in new stack -- Called z2 -- Call accepted by 192.168.0.202 (format ulaw) -- Format for call is ulaw -- IAX2/z2/2 is ringing -- IAX2/z2/2 answered IAX2/[EMAIL PROTECTED]/1 -- Attempting native bridge of IAX2/[EMAIL PROTECTED]/1 and IAX2/z2/2 -- Channel 'IAX2/[EMAIL PROTECTED]/1' ready to transfer -- Channel 'IAX2/z2/2' ready to transfer -- Releasing IAX2/z2/2 and IAX2/[EMAIL PROTECTED]/1 -- Hungup 'IAX2/z2/2' == Spawn extension (geograph, 202, 1) exited non-zero on 'IAX2/[EMAIL PROTECTED]/1' -- Hungup 'IAX2/[EMAIL PROTECTED]/1' -- Registered 'z1' (AUTHENTICATED) at 192.168.0.201:4569 -- Registered 'z2' (AUTHENTICATED) at 192.168.0.202:4569 *CLI *** Rich! Carlos! - Pizzas on me when you come to Cape Town. (or I get to where you are - where's that?) What a learning curve - big thanks and let me give you a suggestion: Take leave the week after next - I'm going to be plugging in 2 internal ISDN BRI cards ;-) (next week will be to sort out the choppy sound to move from my SuSE 9.3 toy box to the production FC3 (dare I try FC4) box - so maybe take next week off as well :-) (Carlos - I'll respond to your zaprtc query later today) Cheers thanks - sincerely hope to be able to return the effort one day. regards to all, Zoltan Rich Adamson wrote: all0w=ulaw all0w=alaw all0w=gsm Look closely at the above four lines. In the allow statement, that appears to be a zero. Change that to allow. Also, I don't know which codecs the phone supports, but you might start playing with disallow=all allow=ulaw and go from there. you're 100% right - I saw the typo when the lines were commented out and the codecs were in the [z1] section. I then changed back in order to shorten the iax.conf file but forgot about the typos. Thanks - it could've taken many more hours for me to notice them again :-) [z1] type=friend user=z1 secret=z1 context=geograph host=dynamic dtmfmode=rfx2833 If you look at /usr/src/asterisk/configs/iax.conf.sample, you'll find that dtmfmode=rfx2833 is not a valid iax statement. Plus its spelled wrong (its rfc2833). Remove it, but add it into your sip.conf if you're going to play with sip. jeeze - dislexia rulz (never change a config file when in a hurry to do something else) *** asterisks response as I dial Asterisk Ready. *CLI iax2 show p peers provisioning *CLI iax2 show peers Name/UsernameHost Mask Port Status z2 192.168.0.202 (D) 255.255.255.255 4569 Unmonitored z1 192.168.0.201 (D) 255.255.255.255 4569 Unmonitored *CLI iax2 show users Username SecretAuthen Def.Context A/C z2 z2003 geograph No z1 z1003 geograph No *CLI -- Accepting AUTHENTICATED call from 192.168.0.201, requested format = 4, actual format = 256 Here is the key: That is telling you it can't find a compatible codec to allow the call to complete. That's the basis for the comments above about the allow=ulaw. *-- Executing Dial(IAX2/[EMAIL PROTECTED]/3, IAX/z2|20|tr) in new stack* Note the above IAX. I think that should be IAX2, so look in your extensions.conf for a dial statement that looks like Dial(IAX/ and change it to Dial(IAX2/. Yep - this too would have taken me a while to notice. ___ 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] Modifying astcc
Hi:
Thank you steve for your help. You saved me time and
headache.
chawki hammoud wrote:
Hi:
Astcc is working fine, except for one thing. It
doesn't give the called party enough time to answer
the phone. If nobody picks up in two rings, astcc
reports back no answer and hangs-up. The only
instant
NOANSWER value was mentioned in astcc.agi script
is:
elsif ($res eq NOANSWER) {
$res =
mystreamfile(astcc-noanswer);
Please help me find what and where to change to
control the time astcc give to the called party to
answer.
Regards;
Chawki Hammoud
Look for the line or lines that call Dial - there's
a number in
there
which is the amount of time to allow for the call to
be answered. You
can
make that longer!
Steve
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[Asterisk-Users] Mitel 5220 Hold?
Anyone had any luck configuring the hold button successfully on a Mitel 5220 SIP phone? Xfer works great. Call completion works fine both directions. Audio has been beautiful. With a working system hold button, this would be a really usable phone. The hold button does make the call appear to be on hold: the light is blinking, the other party is still silently connected, I can pick up a call on another line presence. But I can't recover the call. If I press the line button again, it just hangs up. ___ 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 Realtime database Problem
Hi All, I am facing a problem with makeing asterisk work realtime with mysql, after following the tiki steps which are: uncommented the lines sipuser and sippeers from extconfig.conf copied the res_mysql.conf and configured it with the right parameters checked that mysql is working added the realtime switch to the extensions.conf Now when asterisk is starting I don't see it even to attempt to parse the res_mysql.conf file so I am assuming that there is something missing what is it I don't know. -- Thx MAG ___ 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] phantom incomming calls from asterisk
I have no telco line, only IAX2 to another asterisk server and one SIP phone. David On Sat, 09 Jul 2005 15:40:02 -0400 John Novack [EMAIL PROTECTED] wrote: Many telcos do an automated once a day or once a week or ?? line test, which can appear as an incoming call to some devices. If you unplug your telco line and the events disappear, perhaps that is what is happening? John Novack John Millican wrote: About once a day I have noticed a phantom incoming call with a caller ID of [EMAIL PROTECTED]cut off. When I answer the call there is a dial tone and the call is disconnected. Any clues? David Koski David and List, I am having the same problem. I have an * box at my house with 1 zap (pstn on a X100p clone from digit networks) channel and one sip(linksys ATA). I am getting ring on the ATA but there is no call comming in from the pstn. The following is the CLI output when this happens. I know that there is no call on the pstn because i have an emergency phone(frequent power outages) still connected to the PSTN parallel to the * box and it never rings. All the SIP stuff is on an internal lan only. I only call out on PSTN since all I have available here in nowheare land is dial up :-( All work flawlessly except for this one problem. - Starting simple switch on 'Zap/1-1' Jul 8 13:49:23 NOTICE[6150]: chan_zap.c:5405 ss_thread: Got event 2 (Ring/Answered)... -- Executing Dial(Zap/1-1, sip/677|35) in new stack -- Called 677 -- SIP/677-55a8 is ringing == Spawn extension (default, s, 1) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' Is there anything for Zap like sip debug? My first guess is that I am getting some sort of blip in ring voltage on the PSTN but have no way to prove this. As a posible logic check I unplugged from PSTN, which put zap into Red alarm of course, and then i get no phantom calls. Is there something in the zap driver that shuts down when in red alarm? Any Ideas? John M ___ 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] Cepstral
I have been reading about Cepstral, their voices and the Digium partner agreement with them. I see where they sell the voices and the licenses for them, but what I can't find is how to buy or get Swift? If I understand correctly, swift is the actual program that makes the speech? Strangely, the Cepstral web site does not explain this... Can someone shed some light? Thanks... ___ 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] chan_capi ASTCC trouble
Hi all I am wondering if anyone has had a similar trouble to this: The timeout arguments in the dial command does not work. The caller does not get disconnected when the timeout reaches zero. I am not sure if this is a chan_capi issue, or a asterisk issue. I am using CVS-head and chan_capi CVS head also. Any suggestions or help will be appreciated. Thanks Clive ___ 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] IAXphone - ip address - extension number.
Carlos Alperin wrote: Zoltan, If you don't mind, can you explain me a little more the ztcfg problem. My experience with that is null, but I need to setup a box for testing purposes only, and I don't want to but a TDM card only for make it work. I know people that has asterisk running on SIP that don't use zaptel Zapata just they do sip also to their provider (which is not my my case) but I never did that. So I plan to install it with Ztdummy on top of FC2. I only tried once to make zaprtc - this was whilst ztdummy was not loading for me. I'll go through what I did in a later email. For now, the ztdummy thing: FC2 is 2.4 and not 2.6 kernel. I remember seeing somewhere that ztdummy reacts better on 2.6 kernels. (is this true anyone???) The thing that killed my ztdummy was that, at make time, I did not have/see/notice and udev errors (so although having read it, I ignored the README.udev file), however whenever I modprobed zaptel before modprobe ztdummy (this is the order it must be done in), it would not load. gl0:/home/zls # modprobe zaptel gl0:/home/zls # lsmod | grep z Module Size Used by zaptel239620 0 crc_ccitt 6144 1 zaptel gl0:/home/zls # modprobe ztdummy Notice: Configuration file is /etc/zaptel.conf :-( line 142: Unable to open master device '/dev/zap/ctl' gl0:/home/zls # lsmod | grep z Module Size Used by ztdummy 7744 0 zaptel239620 1 ztdummy crc_ccitt 6144 1 zaptel gl0:/home/zls # Tzafrir Cohen picked up this error on /dev and pointed me to /usr/src/zaptel-1.0.9/README.udev and once I had dumped those 6 lines into /etc/udev/rules.d/50-udev.rules the modprobe sequence worked. You need to find out if FC2 uses udev or not. (I'll re-run my attempts at zaprtc sort out the emails I got - and email you again just now) Cheers, Zoltan Regarding Ethereal, you should do tcpdump -I ethx (x=number of the port) -W filename, when you finish close it with Ctrl-z, and then you can see the file on the Asterisk or move it to another computer with Etherreal and open it (That is the way I do, so I see what Asterisk gets). Have a great weekend. Carlos ___ 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] Remote SIP Connection using Asterisk // Cisco7940's
Asterisk/phones work perfectly within our LAN. Asterisk box has a public IP - no NAT or firewalls. When I take the phones to a remote location (again, public IP - no NAT or firewalls that I know of) the outgoing audio does not work. I can hear the other party, my phones ring, I can dial out, etc, but the other party cannot hear me (even if I dial #'s, etc). Any ideas? Thanks, I had the same problem with a connection over our local Cable Co., even the engineers could not tell me why. I was able to route around it by putting in a direct route, anything that went through their gateway didn't work as described. Some internet routers and gateways drop rtp, I think, expecially systems designed to filter the traffic. -- Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 Cell: 264-235-5670 Yahoo IM: [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] Cepstral
what I can't find is how to buy or get Swift? If I understand correctly, swift is the actual program that makes the speech? IIRC, you can download everything you need to make the thing talk, including a voice like David. It works exactly like it will when you buy a license except there is some kind of crippling until you install the license key. I don't remember if this is a statement made by the voice each time or a time out. FWIW? I bought that voice and I find it amusing, but not ready for prime time. I had it read articles from a publication and it was ludicrous. I can understand the people talking about ATT, I think I heard a demo that was very convincing. So much depends on what you are trying to do. I just wanted to have a way to allow asterisk to talk in a demo, to show the concept. Unfortunately, showing a talking server with Cepestral's David is little like showing a prototype website: people don't always have the imagination (like we all do here :) to see what this would be like when actually done (or using a better voice in this case). ___ 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] iax.cc opinion request
I am considering using iax.cc (sixtel) and wondering if anyone had opinions, good or bad. Are there outages with any regularity? How responsive are tech support? How is packet loss? I am particularly interested in termination to the UK, but will accept any comments people have. Thanks -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ 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] IAXphone - ip address - extension number.
What a learning curve - big thanks and let me give you a suggestion: Take leave the week after next - I'm going to be plugging in 2 internal ISDN BRI cards ;-) I have never played with a BRI, so won't be able to help on that one. But, there are plenty of folks on the list that have it working. (next week will be to sort out the choppy sound to move from my SuSE 9.3 toy box to the production FC3 (dare I try FC4) box - so maybe take next week off as well :-) I'm using FC3 and it works well. I'd stay away from FC4 for now simply because there are not very many people on this list that have tried it, so help is likely to be almost non-existant. If you try FC3, be sure to read the READMEs in zaptel source directory, etc. There are some additional things you will need to do and it seems a lot of folks miss reading those items. In particular, look for udev stuff. Rich ___ 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] Cepstral
what I can't find is how to buy or get Swift? If I understand correctly, swift is the actual program that makes the speech? IIRC, you can download everything you need to make the thing talk, including a voice like David. It works exactly like it will when you buy a license except there is some kind of crippling until you install the license key. I don't remember if this is a statement made by the voice each time or a time out. FWIW? I bought that voice and I find it amusing, but not ready for prime time. I had it read articles from a publication and it was ludicrous. I can understand the people talking about ATT, I think I heard a demo that was very convincing. So much depends on what you are trying to do. I just wanted to have a way to allow asterisk to talk in a demo, to show the concept. Unfortunately, showing a talking server with Cepestral's David is little like showing a prototype website: people don't always have the imagination (like we all do here :) to see what this would be like when actually done (or using a better voice in this case). I have the emily voice and she sounds much like the marine weather station reports. the crippling is just a message that says it is an unregistered version or the like. Yes you can absolutely tell that it is speech synthesis but it is understandable. You can fiddle with the settings, in the readme this is explained, and make it sound a little better. John M ___ 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] Cepstral
I have been reading about Cepstral, their voices and the Digium partner agreement with them. I see where they sell the voices and the licenses for them, but what I can't find is how to buy or get Swift? If I understand correctly, swift is the actual program that makes the speech? Strangely, the Cepstral web site does not explain this... Can someone shed some light? Thanks... I have been using cepstral for a while now. Swift is the old name(I believe) for cepstral and is placed in the /install_dir/bin directory when you unpack the cepstral download. John M ___ 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] SIPGetHeaders for chan_sip (derived from chan_sip2 )
hello how to drive SIPGetHeaders from chan_sip2 as described in http://www.voip-info.org/tiki-index.php?page=PortaOne+Radius+auth thanks, Kamran Ahmad Sell on Yahoo! Auctions no fees. Bid on great items. http://auctions.yahoo.com/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] GXP-2000 MWI
anyone managed to get MWI going on the GXP-2000 with * CVS-HEAD? I have set up the mailbox in the sip.conf entries but no flashing lights... SIP NOTIFY seems to be being sent out...-- regards, Mark P. EdwardsFWD: 667917 ___ 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] phantom incomming calls from asterisk
About once a day I have noticed a phantom incoming call with a caller ID of [EMAIL PROTECTED]cut off. When I answer the call there is a dial tone and the call is disconnected. Any clues? David Koski David and List, I am having the same problem. I have an * box at my house with 1 zap (pstn on a X100p clone from digit networks) channel and one sip(linksys ATA). I am getting ring on the ATA but there is no call comming in from the pstn. The following is the CLI output when this happens. I know that there is no call on the pstn because i have an emergency phone(frequent power outages) still connected to the PSTN parallel to the * box and it never rings. All the SIP stuff is on an internal lan only. I only call out on PSTN since all I have available here in nowheare land is dial up :-( All work flawlessly except for this one problem. - Starting simple switch on 'Zap/1-1' Jul 8 13:49:23 NOTICE[6150]: chan_zap.c:5405 ss_thread: Got event 2 (Ring/Answered)... -- Executing Dial(Zap/1-1, sip/677|35) in new stack -- Called 677 -- SIP/677-55a8 is ringing == Spawn extension (default, s, 1) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' Is there anything for Zap like sip debug? My first guess is that I am getting some sort of blip in ring voltage on the PSTN but have no way to prove this. As a posible logic check I unplugged from PSTN, which put zap into Red alarm of course, and then i get no phantom calls. Is there something in the zap driver that shuts down when in red alarm? Any Ideas? Try this in zapata.conf for fun: busydetect=yes busycount=6 Let us know if it makes a difference. I have added this to zapata.conf. Will let you know what happens. can you tell me why this might help or point me to a wiki/google? ___ 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] IAX2 softphone for Pocket PC
Does anybody know an IAX2 Softphone for Pocket PC? Ciao Andro ___ 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] Retrieving dtmf, passing to shell, and getting the result
Title: Retrieving dtmf, passing to shell, and getting the result
I have my asterisk server up and running on OS X and now need to add the capability to play a sound file asking for a 5 digit number, play another message asking for a 2 digit number, pass these variables to a shell script, and get the result. I have tried a number of different scenarios but they are not working. I have read through the wiki, past posts, and numerous websites.
The sound files are enter-first enter-second
The shell script is CheckNumbers.sh
exten = 2,2,get_data (enter-first,1,5)
exten = 2,3,get_data (enter-second,1,2)
exten = 2,4,system(/usr/local/Scripts/CheckNumbers.sh ${firstnumber, secondnumber)
I really appreciate your help!
Jane
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Re: [Asterisk-Users] Retrieving dtmf, passing to shell, and getting the result
I have my asterisk server up and running on OS X and now need to add the
capability to play a sound file asking for a 5 digit number, play another
message asking for a 2 digit number, pass these variables to a shell
script, and get the result. I have tried a number of different scenarios
but they are not working. I have read through the wiki, past posts, and
numerous websites.
The sound files are enter-first enter-second
The shell script is CheckNumbers.sh
exten = 2,2,get_data (enter-first,1,5)
exten = 2,3,get_data (enter-second,1,2)
exten = 2,4,system(/usr/local/Scripts/CheckNumbers.sh ${firstnumber,
secondnumber)
I really appreciate your help!
Jane
Jane,
try this
exten = 2,2,read (firstnumber,enter-first,5)
exten = 2,3,read (secondnumber,enter-second,2)
exten = 2,4,system(/usr/local/Scripts/CheckNumbers.sh ${firstnumber}
${secondnumber})
I believe it is the syntax that is holding you back.
John M
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Re: [Asterisk-Users] chan_capi ASTCC trouble
Hi all I am wondering if anyone has had a similar trouble to this: The timeout arguments in the dial command does not work. The caller does not get disconnected when the timeout reaches zero. I am not sure if this is a chan_capi issue, or a asterisk issue. I am using CVS-head and chan_capi CVS head also. Any suggestions or help will be appreciated. Thanks Clive Ok, just did some testing on the dial command using only iax2 and it does disconnect the call, so this may be a chan_capi issue. Any suggestions will be great.:) thanks Clive ___ 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 Realtime database Problem
Did you install res_config_mysql.so from asterisk-addons? -Matthew From: Mohamed A. Gombolaty [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Sun, 10 Jul 2005 12:16:51 +0300 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: [Asterisk-Users] Asterisk Realtime database Problem Hi All, I am facing a problem with makeing asterisk work realtime with mysql, after following the tiki steps which are: uncommented the lines sipuser and sippeers from extconfig.conf copied the res_mysql.conf and configured it with the right parameters checked that mysql is working added the realtime switch to the extensions.conf Now when asterisk is starting I don't see it even to attempt to parse the res_mysql.conf file so I am assuming that there is something missing what is it I don't know. -- Thx MAG ___ 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 properly handle incoming SIP and IAX calls, so user can call back and how to properly make outgoing sip/iax calls through Asterisk ?
Hi, I'm aware that incoming and outgoing calls are going fine when isdn channels are involved - caller id properly identifies calling party, so user can call back But how to properly handle this for iax, sip calls I have few questions : - BTW, what to type for instance in remote firefly to make standalone calls to Asterisk default context or particular extension ? - If I receive incoming sip or iax call and is then saved as for instance in Firefly. Now Firefly would like to call back that caller, but call goes not through Asterisk... Why ? How to do this properly? - Outogoing calls: how to properly send outgoind iax or sip calls through asterisk, so each calling extension gets proper caller id, so can be called back ? Any experience or existing solution to this problem? Any advice ? Regards, Rob. ___ 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] IAXphone - ip address - extension number.
Hi Carlos, OK - I speak from memory and a little bit of newbie fiddling (which thanks to you and Rich took a successful turn). Carlos Alperin wrote: Zoltan, If you don't mind, can you explain me a little more the ztcfg problem. My experience with that is null, but I need to setup a box for testing purposes only, and I don't want to but a TDM card only for make it work. I was under the impression that timing signals can be gotten from digium cards, uhci_usb (using ztdummy) and the rtc (using zaptelrtc). If you follow the thread from http://lists.digium.com/pipermail/asterisk-users/2004-September/063355.html then it suggests that from 2.6 kernel, ztdummy no longer requires USB . zaptelrtc does however require that rtc is *not* built into your kernel so you would have to recompile it without rtc if it is. For me the only ztdummy issue was with udev (see README.udev in the zaptel-1.0.9 folder) and all you would have to do was to check if FC2 has udev or not. Dont forget to modprobe zaptel before modprobe ztdummy before loading asterisk. HTH, Zoltan I know people that has asterisk running on SIP that don't use zaptel Zapata just they do sip also to their provider (which is not my my case) but I never did that. So I plan to install it with Ztdummy on top of FC2. Regarding Ethereal, you should do tcpdump -I ethx (x=number of the port) -W filename, when you finish close it with Ctrl-z, and then you can see the file on the Asterisk or move it to another computer with Etherreal and open it (That is the way I do, so I see what Asterisk gets). Have a great weekend. Carlos ___ 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: MeetMe problem - some parameters ignored
In article [EMAIL PROTECTED], Tony Mountifield [EMAIL PROTECTED] wrote: and also the 'r' option in your other posting, Oops, my mistake. The posting about the 'r' option was from Jason Walker, not Jim Archer. But the same answer still applies. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ 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] IAXphone - ip address - extension number.
If you don't mind, can you explain me a little more the ztcfg problem. My experience with that is null, but I need to setup a box for testing purposes only, and I don't want to but a TDM card only for make it work. I know people that has asterisk running on SIP that don't use zaptel Zapata just they do sip also to their provider (which is not my my case) but I never did that. So I plan to install it with Ztdummy on top of FC2. FWIW, I've been running asterisk on a RHv9 laptop (for client demo purposes), connecting it via iax to our office system, and never worried about ztdummy, etc. Obviously, the laptop has no zap cards. This demo never includes meetme, etc. I also have a backup system (to our office system) running RHv9, and it connects and functions just fine to the primary office system via iax. Sip phones work fine on this backup system as well. That backup system was just recently replaced with an FC3 system, and I have no doubt whatsoever it will function just fine without zap cards although that system has not yet been configured or tested. So, don't be too concerned with making ztdummy (etc) function unless you truly want to support those asterisk apps that need it (eg, meetme and whatever else the wiki points out). ___ 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] Problems with firefly connection via SIP
My firefly softphone is having problems connecting via SIP. When I set it up, one provider does appears to connec, but trying to call results in a 'Couldn't start call' The other responses with a 401 failure code. Xten connects okay via SIP. Is there something about Firefly SIP configuration that I don't know about? IAX connects okay / Obelix This message was sent using IMP, the Internet Messaging Program. ___ 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] phantom incomming calls from asterisk
About once a day I have noticed a phantom incoming call with a caller ID of [EMAIL PROTECTED]cut off. When I answer the call there is a dial tone and the call is disconnected. Any clues? David Koski David and List, I am having the same problem. I have an * box at my house with 1 zap (pstn on a X100p clone from digit networks) channel and one sip(linksys ATA). I am getting ring on the ATA but there is no call comming in from the pstn. The following is the CLI output when this happens. I know that there is no call on the pstn because i have an emergency phone(frequent power outages) still connected to the PSTN parallel to the * box and it never rings. All the SIP stuff is on an internal lan only. I only call out on PSTN since all I have available here in nowheare land is dial up :-( All work flawlessly except for this one problem. - Starting simple switch on 'Zap/1-1' Jul 8 13:49:23 NOTICE[6150]: chan_zap.c:5405 ss_thread: Got event 2 (Ring/Answered)... -- Executing Dial(Zap/1-1, sip/677|35) in new stack -- Called 677 -- SIP/677-55a8 is ringing == Spawn extension (default, s, 1) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' Is there anything for Zap like sip debug? My first guess is that I am getting some sort of blip in ring voltage on the PSTN but have no way to prove this. As a posible logic check I unplugged from PSTN, which put zap into Red alarm of course, and then i get no phantom calls. Is there something in the zap driver that shuts down when in red alarm? Any Ideas? Try this in zapata.conf for fun: busydetect=yes busycount=6 Let us know if it makes a difference. I have added this to zapata.conf. Will let you know what happens. can you tell me why this might help or point me to a wiki/google? Going from memory only (which might be less then accurate), the busycount parameter essentially extends zap detect time. The comments in zapata.conf refer to detecting busy tone, but something from past memory says the parameter affects more then just busy tone detection. The default value is 4 but I've been using 6 or at least a year with an x100p followed by a TDM04b, and I don't have the false ring issues. Sure wish I would have kept a diary of config changes over the last two years rather then rely on memory. It would have been helpful more than once. :( ___ 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] IAXphone - ip address - extension number.
Rich Adamson wrote: If you don't mind, can you explain me a little more the ztcfg problem. My experience with that is null, but I need to setup a box for testing purposes only, and I don't want to but a TDM card only for make it work. I know people that has asterisk running on SIP that don't use zaptel Zapata just they do sip also to their provider (which is not my my case) but I never did that. So I plan to install it with Ztdummy on top of FC2. FWIW, I've been running asterisk on a RHv9 laptop (for client demo purposes), connecting it via iax to our office system, and never worried about ztdummy, etc. Obviously, the laptop has no zap cards. This demo never includes meetme, etc. I'm not actually surprised. In the Makefile you'll see somewhere around line 61 that ztdummy is commented out and, if you *want* ztdummy, you should remove the # before running make linux26. However, if you follow through the use of the variable names, you will see that ztdummy is suffiex anyway so whether you comment it in or out, ztdummy gets compiled. It would me interesting to do an lsmod | grep z on your laptop to see if zaptel ztdummy are loaded. I also have a backup system (to our office system) running RHv9, and it connects and functions just fine to the primary office system via iax. Sip phones work fine on this backup system as well. That backup system was just recently replaced with an FC3 system, and I have no doubt whatsoever it will function just fine without zap cards although that system has not yet been configured or tested. So, don't be too concerned with making ztdummy (etc) function unless you truly want to support those asterisk apps that need it (eg, meetme and whatever else the wiki points out). (I seem to remember IAX needed timing, which is why I was on the ztdummy mission - no cards in my testbox either.) Gotta bounce - chat tomorrow. Cheers, Zoltan ___ 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] Tormenta 2 / E400P cards in AMD 64 bit machines
Hi, Is anyone currently using an E400P card in an AMD 64bit processor machine, running a 64 bit version of Linux? I just tried testing my R2 software with this setup for the first time. The CAS signaling bits are sent and received OK, but so far I seem to get no audio transmitted or received.I am using zaptel 1.0.9 Regards, Steve ___ 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] SMS Handler in Asterisk
Hello all, Recently I migrated all telephony in my house to asterisk thanks to the Asterisk, QuadBRI which works wonderfully well. Some small tweaks to make but that's on the long list. On the short list is the ability to reliable send and receive SMS. For SMS I already built a script email2sms, but sometimes the SMS doesn't get send from some reason, the sms log then reports something like: 2005-07-03T07:56:29 ?OM0C 0 - NUMBER MESSAGE Everytime the Second field contains a ? instead of a Y the sending failed, but asterisk doesn't retry. Next to this, no error in the logs. Am I doing something wrong, or is there an additional app that can fix this/monitors logs and does retries or so? Thanks in advance. -- Met Vriendelijke groet/Yours Sincerely Stijn Jonker [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] IAXphone - ip address - extension number.
If you don't mind, can you explain me a little more the ztcfg problem. My experience with that is null, but I need to setup a box for testing purposes only, and I don't want to but a TDM card only for make it work. I know people that has asterisk running on SIP that don't use zaptel Zapata just they do sip also to their provider (which is not my my case) but I never did that. So I plan to install it with Ztdummy on top of FC2. FWIW, I've been running asterisk on a RHv9 laptop (for client demo purposes), connecting it via iax to our office system, and never worried about ztdummy, etc. Obviously, the laptop has no zap cards. This demo never includes meetme, etc. I'm not actually surprised. In the Makefile you'll see somewhere around line 61 that ztdummy is commented out and, if you *want* ztdummy, you should remove the # before running make linux26. However, if you follow through the use of the variable names, you will see that ztdummy is suffiex anyway so whether you comment it in or out, ztdummy gets compiled. It would me interesting to do an lsmod | grep z on your laptop to see if zaptel ztdummy are loaded. I also have a backup system (to our office system) running RHv9, and it connects and functions just fine to the primary office system via iax. Sip phones work fine on this backup system as well. That backup system was just recently replaced with an FC3 system, and I have no doubt whatsoever it will function just fine without zap cards although that system has not yet been configured or tested. So, don't be too concerned with making ztdummy (etc) function unless you truly want to support those asterisk apps that need it (eg, meetme and whatever else the wiki points out). (I seem to remember IAX needed timing, which is why I was on the ztdummy mission - no cards in my testbox either.) Okay, just fired up the laptop and it registered with our office system just fine using iax2. The laptop is RHv9 with cvs-head from 4/25/05 with no modifications of any sort. The lsmod lists zaptel only, Used = 0, and no ztdummy listed. ___ 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] SMS Handler in Asterisk
On 18:21, Sun 10 Jul 05, Stijn Jonker wrote: Hello all, Recently I migrated all telephony in my house to asterisk thanks to the Asterisk, QuadBRI which works wonderfully well. Some small tweaks to make but that's on the long list. On the short list is the ability to reliable send and receive SMS. For SMS I already built a script email2sms, but sometimes the SMS doesn't get send from some reason, the sms log then reports something like: 2005-07-03T07:56:29 ?OM0C 0 - NUMBER MESSAGE Everytime the Second field contains a ? instead of a Y the sending failed, but asterisk doesn't retry. Next to this, no error in the logs. Am I doing something wrong, or is there an additional app that can fix this/monitors logs and does retries or so? Thanks in advance. Hi, This won't answer your question, sorry. How are you sending SMS ? I'm in NL too, and can't seem to find a way to send SMS with asterisk. The only way I found was some service on the internet that sells SMS credits for asterisk users but it would be nice to know how you are doing it. -- Michiel van Baak http://michiel.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Why is it drug addicts and computer afficionados are both called 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] TDM04B Outbound calls
I just install a Digium TDM04B card. I created 4 separate Zap channels and one outbound routing containing zap channels from 1 to 4. If a phone line is plug in Zap/1 then works fine, but if I unplug phone line from Zap/1 (simulating fail) the system keep dialing out on Zap/1, even with no dial tone; Only if Zap/1 is busy it will use Zap/2. There is any work around or different setting to avoid this situation? I don't believe asterisk has any code to detect whether a pstn line is plugged in or not. The chipset on the TDM-fxo modules do support that function, but the drivers don't do anything with it right now. Mark added code for that about a year ago, but commented it out within a day or two as it caused problems for some folks. Don't know if that code remains in the drivers as yet or not. ___ 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] phantom incomming calls from asterisk
About once a day I have noticed a phantom incoming call with a caller ID of [EMAIL PROTECTED]cut off. When I answer the call there is a dial tone and the call is disconnected. Any clues? David Koski David and List, I am having the same problem. I have an * box at my house with 1 zap (pstn on a X100p clone from digit networks) channel and one sip(linksys ATA). I am getting ring on the ATA but there is no call comming in from the pstn. The following is the CLI output when this happens. I know that there is no call on the pstn because i have an emergency phone(frequent power outages) still connected to the PSTN parallel to the * box and it never rings. All the SIP stuff is on an internal lan only. I only call out on PSTN since all I have available here in nowheare land is dial up :-( All work flawlessly except for this one problem. - Starting simple switch on 'Zap/1-1' Jul 8 13:49:23 NOTICE[6150]: chan_zap.c:5405 ss_thread: Got event 2 (Ring/Answered)... -- Executing Dial(Zap/1-1, sip/677|35) in new stack -- Called 677 -- SIP/677-55a8 is ringing == Spawn extension (default, s, 1) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' Is there anything for Zap like sip debug? My first guess is that I am getting some sort of blip in ring voltage on the PSTN but have no way to prove this. As a posible logic check I unplugged from PSTN, which put zap into Red alarm of course, and then i get no phantom calls. Is there something in the zap driver that shuts down when in red alarm? Any Ideas? Try this in zapata.conf for fun: busydetect=yes busycount=6 Let us know if it makes a difference. I have added this to zapata.conf. Will let you know what happens. can you tell me why this might help or point me to a wiki/google? Going from memory only (which might be less then accurate), the busycount parameter essentially extends zap detect time. The comments in zapata.conf refer to detecting busy tone, but something from past memory says the parameter affects more then just busy tone detection. The default value is 4 but I've been using 6 or at least a year with an x100p followed by a TDM04b, and I don't have the false ring issues. Sure wish I would have kept a diary of config changes over the last two years rather then rely on memory. It would have been helpful more than once. :( Well I added these settings to zapata.conf and am still getting the phantom rings, 2 so far this morning! have been watching ztmonitor and am seeing that that rx audio level is showing a constant ###* with an rx gain setting of -7.5 in zapata.conf. If i set gain much less it gets hard to hear voice from callers. With gain at 0.0 i get * with some peaks above this. Is this normal? John M ___ 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] (no subject)
Im trying to get Asterisk to accept incoming calls from budgetphone.nl. When I dial my budgetphone nr on a PSTN KPN line it immediately gives a busy tone. I tried X-lite, which worked perfect, so my modem (with nat) probably is not the problem. I did a sip debug and got the following output. Because Im new to Asterisk I cant get the error why this is not working. To me it all looks fine, no warnings or what so ever The settings in sip.conf and extensions.conf are identical to those of http://www.voip-info.org/tiki-index.php?page=Talkin2ya Does anyone know what Im doing wrong Thanks, Peter. --- output of sip debug --- 11 headers, 0 lines Reliably Transmitting (no NAT) to 81.23.228.150:5060: REGISTER sip:budgetphone.nl SIP/2.0 Via: SIP/2.0/UDP 192.168.2.3:5060;branch=z9hG4bK732b82fc From: sip:[EMAIL PROTECTED];tag=as5dc83db4 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 REGISTER User-Agent: Asterisk PBX Expires: 120 Contact: sip:[EMAIL PROTECTED] Event: registration Content-Length: 0 --- server*CLI -- SIP read from 81.23.228.150:5060: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.2.3:5060;branch=z9hG4bK732b82fc From: sip:[EMAIL PROTECTED];tag=as5dc83db4 To: sip:[EMAIL PROTECTED];tag=9b5971f23d18872ff678d4e9dae023f8.247a Call-ID: [EMAIL PROTECTED] CSeq: 102 REGISTER WWW-Authenticate: Digest realm=budgetphone.nl, nonce=42d15009299d7652e8da589cee2723af4b6a96ca Server: Sip EXpress router (0.8.14-5 (i386/linux)) Content-Length: 0 --- (9 headers 0 lines)--- Responding to challenge, registration to domain/host name budgetphone.nl 12 headers, 0 lines Reliably Transmitting (no NAT) to 81.23.228.150:5060: REGISTER sip:budgetphone.nl SIP/2.0 Via: SIP/2.0/UDP 192.168.2.3:5060;branch=z9hG4bK1818547e From: sip:[EMAIL PROTECTED];tag=as7e56000d To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 103 REGISTER User-Agent: Asterisk PBX Authorization: Digest username=31717110342, realm=budgetphone.nl, algorithm=MD5, uri=sip:budgetphone.nl, nonce=42d15009299d7652e8da589cee2723af4b6a96ca, response=cd69279e6a2512fd48d267ceea3394da, opaque= Expires: 120 Contact: sip:[EMAIL PROTECTED] Event: registration Content-Length: 0 --- server*CLI -- SIP read from 81.23.228.150:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.3:5060;branch=z9hG4bK1818547e From: sip:[EMAIL PROTECTED];tag=as7e56000d To: sip:[EMAIL PROTECTED];tag=9b5971f23d18872ff678d4e9dae023f8.98b0 Call-ID: [EMAIL PROTECTED] CSeq: 103 REGISTER Contact: sip:[EMAIL PROTECTED]:5060;q=0.00;expires=120 Server: Sip EXpress router (0.8.14-5 (i386/linux)) Content-Length: 0 --- (9 headers 0 lines)--- Jul 10 18:38:04 NOTICE[26004]: chan_sip.c:8266 handle_response: Outbound Registration: Expiry for budgetphone.nl is 120 sec (Scheduling reregistration in 105000 ms) Destroying call '[EMAIL PROTECTED]' server*CLI -- SIP read from 81.23.228.150:5060: INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Max-Forwards: 10 Record-Route: sip:[EMAIL PROTECTED];ftag=as47419911;lr=on Via: SIP/2.0/UDP 81.23.228.150;branch=z9hG4bK7842.b2602997.0 Via: SIP/2.0/UDP 212.203.28.2:5060;branch=z9hG4bK2de815aa From: 0031172651375 sip:[EMAIL PROTECTED];tag=as47419911 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Sun, 10 Jul 2005 16:37:54 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 345 v=0 o=root 26318 26318 IN IP4 212.203.28.2 s=session c=IN IP4 81.23.228.139 t=0 0 m=audio 36634 RTP/AVP 3 18 5 0 97 110 101 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:110 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- (15 headers 15 lines)--- Using INVITE request as basis request - [EMAIL PROTECTED] Sending to 81.23.228.150 : 5060 (NAT) Found peer '31717110342' Reliably Transmitting (NAT) to 81.23.228.150:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 81.23.228.150;branch=z9hG4bK7842.b2602997.0;received=81.23.228.150;rport=5060 Via: SIP/2.0/UDP 212.203.28.2:5060;branch=z9hG4bK2de815aa From: 0031172651375 sip:[EMAIL PROTECTED];tag=as47419911 To: sip:[EMAIL PROTECTED];tag=as3f35655f Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: sip:[EMAIL PROTECTED] Proxy-Authenticate: Digest realm=asterisk, nonce=555b996d Content-Length: 0 --- Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms server*CLI -- SIP read from 81.23.228.150:5060: ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 81.23.228.150;branch=z9hG4bK7842.b2602997.0 From: 0031172651375 sip:[EMAIL PROTECTED];tag=as47419911 Call-ID: [EMAIL PROTECTED] To: sip:[EMAIL
Re: [Asterisk-Users] TDM04B Outbound calls
If that is the case, then why on one port FXO (WCFXO) it work different. If there is no dial tone on this card system will play All circuit are busy now and if a second card is installed the call will rollover to the second card automatically. My concern is on a 4 line system if the first line loose dial tone nobody can make outgoing calls unless first channel is busy. - Original Message - From: Rich Adamson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, July 10, 2005 1:42 PM Subject: Re: [Asterisk-Users] TDM04B Outbound calls I just install a Digium TDM04B card. I created 4 separate Zap channels and one outbound routing containing zap channels from 1 to 4. If a phone line is plug in Zap/1 then works fine, but if I unplug phone line from Zap/1 (simulating fail) the system keep dialing out on Zap/1, even with no dial tone; Only if Zap/1 is busy it will use Zap/2. There is any work around or different setting to avoid this situation? I don't believe asterisk has any code to detect whether a pstn line is plugged in or not. The chipset on the TDM-fxo modules do support that function, but the drivers don't do anything with it right now. Mark added code for that about a year ago, but commented it out within a day or two as it caused problems for some folks. Don't know if that code remains in the drivers as yet or not. ___ 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] Incoming calls from BudgetPhone.nl
(this time with subject) Hello, Im trying to get Asterisk to accept incoming calls from budgetphone.nl. When I dial my budgetphone nr on a PSTN KPN line it immediately gives a busy tone. I tried X-lite, which worked perfect, so my modem (with nat) probably is not the problem. I did a sip debug and got the following output. Because Im new to Asterisk I cant get the error why this is not working. To me it all looks fine, no warnings or what so ever The settings in sip.conf and extensions.conf are identical to those of http://www.voip-info.org/tiki-index.php?page=Talkin2ya Does anyone know what Im doing wrong Thanks, Peter. --- output of sip debug --- 11 headers, 0 lines Reliably Transmitting (no NAT) to 81.23.228.150:5060: REGISTER sip:budgetphone.nl SIP/2.0 Via: SIP/2.0/UDP 192.168.2.3:5060;branch=z9hG4bK732b82fc From: sip:[EMAIL PROTECTED];tag=as5dc83db4 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 REGISTER User-Agent: Asterisk PBX Expires: 120 Contact: sip:[EMAIL PROTECTED] Event: registration Content-Length: 0 --- server*CLI -- SIP read from 81.23.228.150:5060: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.2.3:5060;branch=z9hG4bK732b82fc From: sip:[EMAIL PROTECTED];tag=as5dc83db4 To: sip:[EMAIL PROTECTED];tag=9b5971f23d18872ff678d4e9dae023f8.247a Call-ID: [EMAIL PROTECTED] CSeq: 102 REGISTER WWW-Authenticate: Digest realm=budgetphone.nl, nonce=42d15009299d7652e8da589cee2723af4b6a96ca Server: Sip EXpress router (0.8.14-5 (i386/linux)) Content-Length: 0 --- (9 headers 0 lines)--- Responding to challenge, registration to domain/host name budgetphone.nl 12 headers, 0 lines Reliably Transmitting (no NAT) to 81.23.228.150:5060: REGISTER sip:budgetphone.nl SIP/2.0 Via: SIP/2.0/UDP 192.168.2.3:5060;branch=z9hG4bK1818547e From: sip:[EMAIL PROTECTED];tag=as7e56000d To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 103 REGISTER User-Agent: Asterisk PBX Authorization: Digest username=31717110342, realm=budgetphone.nl, algorithm=MD5, uri=sip:budgetphone.nl, nonce=42d15009299d7652e8da589cee2723af4b6a96ca, response=cd69279e6a2512fd48d267ceea3394da, opaque= Expires: 120 Contact: sip:[EMAIL PROTECTED] Event: registration Content-Length: 0 --- server*CLI -- SIP read from 81.23.228.150:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.3:5060;branch=z9hG4bK1818547e From: sip:[EMAIL PROTECTED];tag=as7e56000d To: sip:[EMAIL PROTECTED];tag=9b5971f23d18872ff678d4e9dae023f8.98b0 Call-ID: [EMAIL PROTECTED] CSeq: 103 REGISTER Contact: sip:[EMAIL PROTECTED]:5060;q=0.00;expires=120 Server: Sip EXpress router (0.8.14-5 (i386/linux)) Content-Length: 0 --- (9 headers 0 lines)--- Jul 10 18:38:04 NOTICE[26004]: chan_sip.c:8266 handle_response: Outbound Registration: Expiry for budgetphone.nl is 120 sec (Scheduling reregistration in 105000 ms) Destroying call '[EMAIL PROTECTED]' server*CLI -- SIP read from 81.23.228.150:5060: INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Max-Forwards: 10 Record-Route: sip:[EMAIL PROTECTED];ftag=as47419911;lr=on Via: SIP/2.0/UDP 81.23.228.150;branch=z9hG4bK7842.b2602997.0 Via: SIP/2.0/UDP 212.203.28.2:5060;branch=z9hG4bK2de815aa From: 0031172651375 sip:[EMAIL PROTECTED];tag=as47419911 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Sun, 10 Jul 2005 16:37:54 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 345 v=0 o=root 26318 26318 IN IP4 212.203.28.2 s=session c=IN IP4 81.23.228.139 t=0 0 m=audio 36634 RTP/AVP 3 18 5 0 97 110 101 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:110 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- (15 headers 15 lines)--- Using INVITE request as basis request - [EMAIL PROTECTED] Sending to 81.23.228.150 : 5060 (NAT) Found peer '31717110342' Reliably Transmitting (NAT) to 81.23.228.150:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 81.23.228.150;branch=z9hG4bK7842.b2602997.0;received=81.23.228.150;rport=506 0 Via: SIP/2.0/UDP 212.203.28.2:5060;branch=z9hG4bK2de815aa From: 0031172651375 sip:[EMAIL PROTECTED];tag=as47419911 To: sip:[EMAIL PROTECTED];tag=as3f35655f Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: sip:[EMAIL PROTECTED] Proxy-Authenticate: Digest realm=asterisk, nonce=555b996d Content-Length: 0 --- Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms server*CLI -- SIP read from 81.23.228.150:5060: ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 81.23.228.150;branch=z9hG4bK7842.b2602997.0 From: 0031172651375 sip:[EMAIL PROTECTED];tag=as47419911 Call-ID: [EMAIL PROTECTED] To: sip:[EMAIL PROTECTED];tag=as3f35655f CSeq: 102 ACK User-Agent: Sip EXpress router(0.8.14-5
Re: [Asterisk-Users] SMS Handler in Asterisk
Hello Stijn, This is not a problem on the Asterisk website. I have a working KPN SMS setup. Just the KPN SMSC is buggy. Sometimes it does not accept the SMS. Other times it accepts the SMS, but doesnt send it -or- will send it after a LONG delay (couple of hours, sometimes days). Cheers, Rene Kluwen Chimit p.s.: For people wanting details about the setup, email me directly. Sometimes I dont read every message on this list. Hello all, Recently I migrated all telephony in my house to asterisk thanks to the Asterisk, QuadBRI which works wonderfully well. Some small tweaks to make but that's on the long list. On the short list is the ability to reliable send and receive SMS. For SMS I already built a script email2sms, but sometimes the SMS doesn't get send from some reason, the sms log then reports something like: 2005-07-03T07:56:29 ?OM0C 0 - NUMBER MESSAGE Everytime the Second field contains a ? instead of a Y the sending failed, but asterisk doesn't retry. Next to this, no error in the logs. Am I doing something wrong, or is there an additional app that can fix this/monitors logs and does retries or so? Thanks in advance. -- Met Vriendelijke groet/Yours Sincerely Stijn Jonker [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] Cepstral
Thanks William and John, I'll look again for that download. Comments below... --On Sunday, July 10, 2005 1:50 PM +0200 Wilson Pickett [EMAIL PROTECTED] wrote: FWIW? I bought that voice and I find it amusing, but not ready for prime time. I had it read articles from a publication and it was ludicrous. I can understand the people talking about ATT, I think I heard a demo that was very convincing. What is ATT? Is it another text to speech engine? I installed Festival a few days ago and have been playing with it. It sounds okay, but I decided to look to see if I could find something better. Some searching on this list and elsewhere revealed that people were raving about Cepstral, so I figured I would try it. I found their demo page and, honestly, didn't think it sounded much better than Festival. But I like that it had different voice options and Festival seems to have an Irish accent. Not that I mind an Irish accent, but in the US it would not be expected. Is there another product I should be looking at? I don't even know for sure what I am going to do with it yet, but I am certain I'll think of something. This is too cool not to use, but only if it is useful. So much depends on what you are trying to do. I just wanted to have a way to allow asterisk to talk in a demo, to show the concept. Unfortunately, showing a talking server with Cepestral's David is little like showing a prototype website: people don't always have the imagination (like we all do here :) to see what this would be like when actually done (or using a better voice in this case). People can be turned off very quickly. That's exactly why, whatever I end up doing with this, it needs to sound clear and be understandable. No one gives anything a second chance :( Jim ___ 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] IAXphone - ip address - extension number.
Rich Adamson wrote: If you don't mind, can you explain me a little more the ztcfg problem. My experience with that is null, but I need to setup a box for testing purposes only, and I don't want to but a TDM card only for make it work. I know people that has asterisk running on SIP that don't use zaptel Zapata just they do sip also to their provider (which is not my my case) but I never did that. So I plan to install it with Ztdummy on top of FC2. FWIW, I've been running asterisk on a RHv9 laptop (for client demo purposes), connecting it via iax to our office system, and never worried about ztdummy, etc. Obviously, the laptop has no zap cards. This demo never includes meetme, etc. I'm not actually surprised. In the Makefile you'll see somewhere around line 61 that ztdummy is commented out and, if you *want* ztdummy, you should remove the # before running make linux26. However, if you follow through the use of the variable names, you will see that ztdummy is suffiex anyway so whether you comment it in or out, ztdummy gets compiled. It would me interesting to do an lsmod | grep z on your laptop to see if zaptel ztdummy are loaded. I also have a backup system (to our office system) running RHv9, and it connects and functions just fine to the primary office system via iax. Sip phones work fine on this backup system as well. That backup system was just recently replaced with an FC3 system, and I have no doubt whatsoever it will function just fine without zap cards although that system has not yet been configured or tested. So, don't be too concerned with making ztdummy (etc) function unless you truly want to support those asterisk apps that need it (eg, meetme and whatever else the wiki points out). (I seem to remember IAX needed timing, which is why I was on the ztdummy mission - no cards in my testbox either.) Okay, just fired up the laptop and it registered with our office system just fine using iax2. The laptop is RHv9 with cvs-head from 4/25/05 with no modifications of any sort. The lsmod lists zaptel only, Used = 0, and no ztdummy listed. ___ 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 Hmmm - interesting. I did an rmmod on both ztdummy and zaptel and then fired up asterisk. I could still make a call from 1 extension to another, but both music_on_hold *and* IAX2 complained about timing (but I could still make the call!!). I haven't played with MOH yet, so I dont know if the sound will work or be choppy. Let the mystery remain? Cheers, Zoltan. *** [res_musiconhold.so] = (Music On Hold Resource) == Parsing '/etc/asterisk/musiconhold.conf': Found Jul 10 19:20:53 WARNING[3548]: res_musiconhold.c:565 moh_register: Unable to open pseudo channel for timing... Sound may be choppy. == Registered application 'MusicOnHold' == Registered application 'WaitMusicOnHold' == Registered application 'SetMusicOnHold' *** cut a bit out ** [chan_iax2.so] = (Inter Asterisk eXchange (Ver 2)) Jul 10 19:20:53 WARNING[3548]: chan_iax2.c:7477 load_module: Unable to open IAX timing interface: No such file or directory == Manager registered action IAXpeers == Parsing '/etc/asterisk/iax.conf': Found -- Seeding 'z1' at 192.168.0.201:4569 for 60 -- Seeding 'z2' at 192.168.0.202:4569 for 60 == Registered channel type 'IAX2' (Inter Asterisk eXchange Driver (Ver 2)) == Using TOS bits 0 == IAX Ready and Listening on 0.0.0.0 port 4569 == Loaded firmware 'iaxy.bin' == Parsing '/etc/asterisk/iaxprov.conf': Found -- Loaded provisioning template 'default' *** ___ 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] phantom incomming calls from asterisk
I have had this same problem for quite some time on one of my setups. Everyday around the same time I get a phantom ring. If I'm at the location I pick up the sip phone asterisk rings and hangup. Asterisk hangs up the zap line...then answers the zap line again. This happens anywhere from 2-4 times in a row. If I am not at the location asterisk keeps the line off hook for an extended period of time, enough to where I hear a warning from my telco If you'd like to make a call please hangup and try again ( I noticed this from listening to recordings,all incoming calls are recorded ). At 3 other locations this has never happened. I previously had busydetect=yes and busycount=4. After seeing the response by Rich I changed busycount to 6. The problem remains. I'm going to try an even higher busycount number and hope things clear up. ~Jeremy ___ 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] Incoming calls from BudgetPhone.nl
On 19:16, Sun 10 Jul 05, Peter Raaijmaker wrote: (this time with subject) Hello, I?m trying to get Asterisk to accept incoming calls from budgetphone.nl. When I dial my budgetphone nr on a PSTN KPN line it immediately gives a busy tone. I tried X-lite, which worked perfect, so my modem (with nat) probably is not the problem. I did a sip debug and got the following output. Because I?m new to Asterisk I can?t get the error why this is not working. To me it all looks fine, no warnings or what so ever? ? The settings in sip.conf and extensions.conf are identical to those of http://www.voip-info.org/tiki-index.php?page=Talkin2ya ? Does anyone know what I?m doing wrong ? Can you show us the relevant part in sip.conf and extensions.conf. It is working fine here (cept for audio quality and stability of the sip registration, I'm trashing them soon) If you post it I can compare it with my setup and maybe that will show us what's going wrong on your setup -- Michiel van Baak http://michiel.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Why is it drug addicts and computer afficionados are both called 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] iax fwd - calling twice
Hi, testing a new fwd account, dialling from sip4030 to my FWD number, sip4021 rings as defined in extensions conf. Why is this happening twice? -- Executing SetCallerID(SIP/4030-a7f2, HTCAS) in new stack -- Executing Dial(SIP/4030-a7f2, IAX2/617533:[EMAIL PROTECTED]/617533|60|r) in new stack -- Called 617533:[EMAIL PROTECTED]/617533 -- Call accepted by 65.39.205.121 (format ulaw) -- Format for call is ulaw -- Accepting AUTHENTICATED call from 65.39.205.121, requested format = 4, actual format = 4 -- Executing Dial(IAX2/[EMAIL PROTECTED]:4569/3, sip/4021|15|r) in new stack -- Called 4021 -- Accepting AUTHENTICATED call from 65.39.205.121, requested format = 4, actual format = 4 -- Executing Dial(IAX2/[EMAIL PROTECTED]:4569/4, sip/4021|15|r) in new stack -- Called 4021 -- IAX2/65.39.205.121:4569/1 is ringing -- SIP/4021-c034 is ringing -- SIP/4021-717d is ringing -- Hungup 'IAX2/65.39.205.121:4569/1' ___ 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] GXP-2000 MWI
On 10/07/05, Mark Edwards [EMAIL PROTECTED] wrote: anyone managed to get MWI going on the GXP-2000 with * CVS-HEAD? I have set up the mailbox in the sip.conf entries but no flashing lights... SIP NOTIFY seems to be being sent out... Yes, works like a charm here. Firmware 1.0.1.9. Peter -- Peter Bowyer Email: [EMAIL PROTECTED] Tel: +44 1296 768003 VoIP: sip:[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] DELL 2800 : PCI Parity error
Still not resolved On Wed, 2005-06-08 at 01:16, David John Walsh wrote: Frank Did you ever resolve this? If so what was the issue? On 03/05/05, list [EMAIL PROTECTED] wrote: Hi, I am struggling to get rid of a conflict on DELL 2800 : PCI Parity error (EB113 on the display) I am learning linux and asterisk as I go along, there might be obvious things I should know, but bear with me. From demsg below my 2 digium cards installed are listed (no config or connections done to digium cards yet), the conflict is with the TDM400P card, without that card, in any slot, no alarm. Zapata Telephony Interface Registered on major 196 Registered Tormenta2 PCI Controller version: 24 FALC version: TE110P: Setting up global serial parameters for E1 FALC V1.2 TE110P: Successfully initialized serial bus for card Found a Wildcard: Digium Wildcard TE110P T1/E1 Freshmaker version: 71 Freshmaker passed register test Uhhuh. NMI received. Dazed and confused, but trying to continue You probably have a hardware problem with your RAM chips Module 0: Installed -- AUTO FXS/DPO Module 1: Not installed Module 2: Not installed Module 3: Installed -- AUTO FXO (FCC mode) Found a Wildcard TDM: Wildcard TDM400P REV E/F (4 modules) Registered tone zone 8 (Norway) TE110P: Span configured for CCS/HDB3/CRC4 Calling startup (flags is 4099) wcte1xxp: Setting yellow alarm usb.c: registered new driver wcusb Wildcard USB FXS Interface driver registered TE110P: Span configured for CCS/HDB3/CRC4 Calling startup (flags is 4099) Registered tone zone 8 (Norway) TE110P: Span configured for CCS/HDB3/CRC4 Calling startup (flags is 4099) Registered tone zone 8 (Norway) ramchip problem is false, without the card all ok, ramtests on machine as well. lsmod shows wcusb driver on zaptel, I dont need that, can I remove it? is that a problem or not? # lsmod Module Size Used byNot tainted usbserial 23964 0 (autoclean) (unused) lp 9156 0 (autoclean) parport38848 0 (autoclean) [lp] autofs416984 0 (autoclean) (unused) wcusb 19552 0 (unused) wctdm 41088 0 (unused) wcte11xp 22048 0 (unused) zaptel182080 4 [wcusb wctdm wcte11xp] e1000 77884 1 (autoclean) floppy 57552 0 (autoclean) sg 37388 0 (autoclean) microcode 6912 0 (autoclean) ide-cd 34016 0 (autoclean) cdrom 32896 0 (autoclean) [ide-cd] keybdev 2976 0 (unused) mousedev5688 1 hid22308 0 (unused) input 6176 0 [keybdev mousedev hid] ehci-hcd 20776 0 (unused) usb-uhci 26860 0 (unused) usbcore81152 1 [usbserial wcusb hid ehci-hcd usb-uhci] ext3 89960 6 jbd55060 6 [ext3] megaraid2 38344 7 diskdumplib 5228 0 [megaraid2] sd_mod 13904 14 scsi_mod 115112 2 [sg megaraid2 sd_mod] finally my interrupts, bit confusing to me, looks like I have dual processor, can see the NMI but what else can be found here? # cat /proc/interrupts CPU0 CPU1 0:32983953303167IO-APIC-edge timer 1: 3300 2876IO-APIC-edge keyboard 2: 0 0 XT-PIC cascade 8: 0 1IO-APIC-edge rtc 12: 236637 237965IO-APIC-edge PS/2 Mouse 14: 261779 262965IO-APIC-edge ide0 16: 0 0 IO-APIC-level usb-uhci 18: 0 0 IO-APIC-level usb-uhci 19: 0 0 IO-APIC-level usb-uhci 23: 0 24 IO-APIC-level ehci-hcd 29: 33133540 32846566 IO-APIC-level t1xxp 38: 72500 83317 IO-APIC-level megaraid 58: 32838989 33150525 IO-APIC-level wctdm 72: 222855 12 IO-APIC-level eth0 NMI: 1 0 LOC:66014626601460 ERR: 0 MIS: 0 any suggestions from someone
RE: [Asterisk-Users] Incoming calls from BudgetPhone.nl
Rene, I believe you're right, when I disable x-ten's stun server my call isn't coming through anymore. But now I don't have a solution but an extra problem I'm afraid! How to make asterisk run with a stun server? Do I have to set one up myself or can I use the x-ten server for example? Or is there a better way to setup asterisk or my router? Thanks for your help, hopefully you can help me some more! Peter Raaijmakers. -Oorspronkelijk bericht- Van: Rene Kluwen [mailto:[EMAIL PROTECTED] Verzonden: zondag 10 juli 2005 19:28 Aan: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Onderwerp: Re: [Asterisk-Users] Incoming calls from BudgetPhone.nl Long short, Maybe X-Ten has an stun relay setup and Asterisk doesn't? Rene Kluwen Chimit (this time with subject) Hello, Im trying to get Asterisk to accept incoming calls from budgetphone.nl. When I dial my budgetphone nr on a PSTN KPN line it immediately gives a busy tone. I tried X-lite, which worked perfect, so my modem (with nat) probably is not the problem. I did a sip debug and got the following output. Because Im new to Asterisk I cant get the error why this is not working. To me it all looks fine, no warnings or what so ever The settings in sip.conf and extensions.conf are identical to those of http://www.voip-info.org/tiki-index.php?page=Talkin2ya Does anyone know what Im doing wrong Thanks, Peter. --- output of sip debug --- 11 headers, 0 lines Reliably Transmitting (no NAT) to 81.23.228.150:5060: REGISTER sip:budgetphone.nl SIP/2.0 Via: SIP/2.0/UDP 192.168.2.3:5060;branch=z9hG4bK732b82fc From: sip:[EMAIL PROTECTED];tag=as5dc83db4 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 REGISTER User-Agent: Asterisk PBX Expires: 120 Contact: sip:[EMAIL PROTECTED] Event: registration Content-Length: 0 --- server*CLI -- SIP read from 81.23.228.150:5060: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.2.3:5060;branch=z9hG4bK732b82fc From: sip:[EMAIL PROTECTED];tag=as5dc83db4 To: sip:[EMAIL PROTECTED];tag=9b5971f23d18872ff678d4e9dae023f8.247a Call-ID: [EMAIL PROTECTED] CSeq: 102 REGISTER WWW-Authenticate: Digest realm=budgetphone.nl, nonce=42d15009299d7652e8da589cee2723af4b6a96ca Server: Sip EXpress router (0.8.14-5 (i386/linux)) Content-Length: 0 --- (9 headers 0 lines)--- Responding to challenge, registration to domain/host name budgetphone.nl 12 headers, 0 lines Reliably Transmitting (no NAT) to 81.23.228.150:5060: REGISTER sip:budgetphone.nl SIP/2.0 Via: SIP/2.0/UDP 192.168.2.3:5060;branch=z9hG4bK1818547e From: sip:[EMAIL PROTECTED];tag=as7e56000d To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 103 REGISTER User-Agent: Asterisk PBX Authorization: Digest username=31717110342, realm=budgetphone.nl, algorithm=MD5, uri=sip:budgetphone.nl, nonce=42d15009299d7652e8da589cee2723af4b6a96ca, response=cd69279e6a2512fd48d267ceea3394da, opaque= Expires: 120 Contact: sip:[EMAIL PROTECTED] Event: registration Content-Length: 0 --- server*CLI -- SIP read from 81.23.228.150:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.3:5060;branch=z9hG4bK1818547e From: sip:[EMAIL PROTECTED];tag=as7e56000d To: sip:[EMAIL PROTECTED];tag=9b5971f23d18872ff678d4e9dae023f8.98b0 Call-ID: [EMAIL PROTECTED] CSeq: 103 REGISTER Contact: sip:[EMAIL PROTECTED]:5060;q=0.00;expires=120 Server: Sip EXpress router (0.8.14-5 (i386/linux)) Content-Length: 0 --- (9 headers 0 lines)--- Jul 10 18:38:04 NOTICE[26004]: chan_sip.c:8266 handle_response: Outbound Registration: Expiry for budgetphone.nl is 120 sec (Scheduling reregistration in 105000 ms) Destroying call '[EMAIL PROTECTED]' server*CLI -- SIP read from 81.23.228.150:5060: INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Max-Forwards: 10 Record-Route: sip:[EMAIL PROTECTED];ftag=as47419911;lr=on Via: SIP/2.0/UDP 81.23.228.150;branch=z9hG4bK7842.b2602997.0 Via: SIP/2.0/UDP 212.203.28.2:5060;branch=z9hG4bK2de815aa From: 0031172651375 sip:[EMAIL PROTECTED];tag=as47419911 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Sun, 10 Jul 2005 16:37:54 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 345 v=0 o=root 26318 26318 IN IP4 212.203.28.2 s=session c=IN IP4 81.23.228.139 t=0 0 m=audio 36634 RTP/AVP 3 18 5 0 97 110 101 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:110 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- (15 headers 15 lines)--- Using INVITE request as basis request - [EMAIL PROTECTED] Sending to 81.23.228.150 : 5060 (NAT) Found peer '31717110342' Reliably Transmitting (NAT) to 81.23.228.150:5060: SIP/2.0 407
Re: [Asterisk-Users] Incoming calls from BudgetPhone.nl
Try using insecure=very in your peer definition. That makes asterisk not require authentication from your peer if comes from the ip address give in host=xxx.xxx.xxx.xx directive. That helped me receiving calls from my sip provider, which had exactly the same problem. Julian. On 7/10/05, Peter Raaijmaker [EMAIL PROTECTED] wrote: (this time with subject) Hello, I'm trying to get Asterisk to accept incoming calls from budgetphone.nl. When I dial my budgetphone nr on a PSTN KPN line it immediately gives a busy tone. I tried X-lite, which worked perfect, so my modem (with nat) probably is not the problem. I did a sip debug and got the following output. Because I'm new to Asterisk I can't get the error why this is not working. To me it all looks fine, no warnings or what so ever… The settings in sip.conf and extensions.conf are identical to those of http://www.voip-info.org/tiki-index.php?page=Talkin2ya Does anyone know what I'm doing wrong Thanks, 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] Incoming calls from BudgetPhone.nl
Julian, Thanks for your suggestion. The insecure option was already in. Atleast almost; I misspelled it. I wrote insecurity ! Oooops!! Thanks all! Peter Raaijmakers. -Oorspronkelijk bericht- Van: Julian J. M. [mailto:[EMAIL PROTECTED] Verzonden: zondag 10 juli 2005 20:20 Aan: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Onderwerp: Re: [Asterisk-Users] Incoming calls from BudgetPhone.nl Try using insecure=very in your peer definition. That makes asterisk not require authentication from your peer if comes from the ip address give in host=xxx.xxx.xxx.xx directive. That helped me receiving calls from my sip provider, which had exactly the same problem. Julian. On 7/10/05, Peter Raaijmaker [EMAIL PROTECTED] wrote: (this time with subject) Hello, I'm trying to get Asterisk to accept incoming calls from budgetphone.nl. When I dial my budgetphone nr on a PSTN KPN line it immediately gives a busy tone. I tried X-lite, which worked perfect, so my modem (with nat) probably is not the problem. I did a sip debug and got the following output. Because I'm new to Asterisk I can't get the error why this is not working. To me it all looks fine, no warnings or what so ever. The settings in sip.conf and extensions.conf are identical to those of http://www.voip-info.org/tiki-index.php?page=Talkin2ya Does anyone know what I'm doing wrong Thanks, 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] Re: Asterisk-Users Digest, Vol 12, Issue 63
Hello ALL When i dial out to outside, how to detected remote call offhook. i append in extensions.conf [ext-callout];exten = s,1,NVLineDetect(60,d);exten = s,1,NVLineDetect;exten = s,1,NVBackgroundDetect(custom/aa_1);exten = s,1,MachineDetect(7000,2,2200);exten = s,1,NVFaxDetect(10) exten = s,1,NoOp exten = s,2,background(custom/aa_2)exten = s,3,Hangup exten = answer,1,background(custom/aa_11)exten = answer,2,hangup exten = fax,1,txfax(/testfax.tif|9080718)exten = fax,2,Hangup ;exten = fax,1,background(custom/aa_21);exten = fax,2,hangup exten = talk,1,background(custom/aa_12)exten = talk,2,hangup exten = i,1,hangupexten = h,1,hangup and make a new call file test.call Channel: Vpb/g1/2442790MaxRetries: 0WaitTime: 20Context: ext-calloutExtension: sPriority: 1 when i copy test.call to /var/spool/asterisk/outgoing. my asterisk dial out. then play exten = s,2,background(custom/aa_2)? please help me to configure to regconize remote call offhook, then i must be playing wave file. Thanks My hardware is using Voicetronix Openswitch 12, disconnect tone is good. ___ 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] Incoming calls from BudgetPhone.nl
Same here, Audio quality is ok. SIP registration sucks. The helpdesk makes me believe that it is a problem on my side: Your asterisk doesn't respond to a sip request in time. But I have no problems with any other provider, except with Budgetphone. I am not even getting a SIP request, so how do I respond to it? Rene Kluwen Chimit On 19:16, Sun 10 Jul 05, Peter Raaijmaker wrote: (this time with subject) Hello, I?m trying to get Asterisk to accept incoming calls from budgetphone.nl. When I dial my budgetphone nr on a PSTN KPN line it immediately gives a busy tone. I tried X-lite, which worked perfect, so my modem (with nat) probably is not the problem. I did a sip debug and got the following output. Because I?m new to Asterisk I can?t get the error why this is not working. To me it all looks fine, no warnings or what so ever? ? The settings in sip.conf and extensions.conf are identical to those of http://www.voip-info.org/tiki-index.php?page=Talkin2ya ? Does anyone know what I?m doing wrong ? Can you show us the relevant part in sip.conf and extensions.conf. It is working fine here (cept for audio quality and stability of the sip registration, I'm trashing them soon) If you post it I can compare it with my setup and maybe that will show us what's going wrong on your setup -- Michiel van Baak http://michiel.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Why is it drug addicts and computer afficionados are both called 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] Incoming calls from BudgetPhone.nl
On 20:42, Sun 10 Jul 05, Rene Kluwen wrote: Same here, Audio quality is ok. SIP registration sucks. The helpdesk makes me believe that it is a problem on my side: Your asterisk doesn't respond to a sip request in time. But I have no problems with any other provider, except with Budgetphone. I am not even getting a SIP request, so how do I respond to it? Rene Kluwen Chimit Yeah, I tried to explain it to them too several times. They tell me the same story as they told you, and me too doesn't see any requests. The audio is ok on 80% of the calls, the other 20% is plain horrible. Delays, echo, disconnect in middle of call. I don't have those issues with nikotel or iax2 provider. I'm now waiting for my new phone# with iax provider and when that is done I will remove the budgetphone config from my system. -- Michiel van Baak http://michiel.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Why is it drug addicts and computer afficionados are both called 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] Remote SIP Connection using Asterisk // Cisco7940's
Have you tried setting the extension to nat=yes in Asterisk, and then enabling the NAT Enable and NAT Received Processing Enable on the phones? There may be a NAT in between that you do not know about, or it might help for unknown reasons, but it certainly cannot hurt to try! From my experience, you do not want to specify a NAT WAN Address for the phone, because that will limit you to only one working connection behind the NAT. Again, you might have no NAT, but it can't hurt. This raises another question I have: Is there any reason to NOT specify nat=yes for all of your extensions, even if they are not behind NAT? From what I can tell, it will help if the phone is indeed behind a NAT, and will not hurt if it isn't (I assume that I must be missing something here...) Tom On Jul 9, 2005, at 7:33 PM, Carlos Alperin wrote: Some way you should have a udp filter between you box and your phones. I see that before. Can you call those phones? Carlos Alperin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ross Overstreet Sent: Saturday, July 09, 2005 2:04 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Remote SIP Connection using Asterisk // Cisco7940's Asterisk/phones work perfectly within our LAN. Asterisk box has a public IP - no NAT or firewalls. When I take the phones to a remote location (again, public IP - no NAT or firewalls that I know of) the outgoing audio does not work. I can hear the other party, my phones ring, I can dial out, etc, but the other party cannot hear me (even if I dial #'s, etc). Any ideas? Thanks, Ross ___ 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] iax.cc opinion request
On 7/10/2005, trixter wrote: I am considering using iax.cc (sixtel) and wondering if anyone had opinions, good or bad. Are there outages with any regularity? How responsive are tech support? How is packet loss? I am particularly interested in termination to the UK, but will accept any comments people have. Well - here we have a quandary. Opinion? Bad. (But so good) No outages that I can place on Sixtel - 24/7 rock solid - think a router hiccupped once for a couple of hours, but it wasn't theirs. Packet loss - again - as good or better that cell phones. Can't fault them (or him) there. UK termination (DID?) - can't say - thought they (or him) were US only. Tech support? Hahahahahahahahahahaha Ouch - my sides hurt! Took a month to get a DID. Still two months gone waiting for a Custom Toll Free - the primary reason I went with Sixtel... Though they take the money from your account as soon as you order. But then they don't continue billing until whatever you ordered works. It just takes SO long to get it working. Brett ___ 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_capi ASTCC trouble
On Sun, 10 Jul 2005, Clive wrote: Hi all I am wondering if anyone has had a similar trouble to this: The timeout arguments in the dial command does not work. The caller does not get disconnected when the timeout reaches zero. I am not sure if this is a chan_capi issue, or a asterisk issue. I am using CVS-head and chan_capi CVS head also. Any suggestions or help will be appreciated. Thanks Clive Ok, just did some testing on the dial command using only iax2 and it does disconnect the call, so this may be a chan_capi issue. As far as I know, the timeout and hangup logic is done within Asterisk e.g. dial-application. chan-capi does not know anything about a timeout, so I don't know how this can be the location of the problem. Armin ___ 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] SMS Handler in Asterisk
Hello Michiel,
On 10-Jul-2005 18:43, Michiel van Baak wrote:
On 18:21, Sun 10 Jul 05, Stijn Jonker wrote:
Hello all,
On the short list is the ability to reliable send and receive SMS.
For SMS I already built a script email2sms, but sometimes the SMS
doesn't get send from some reason, the sms log then reports something like:
2005-07-03T07:56:29 ?OM0C 0 - NUMBER MESSAGE
Everytime the Second field contains a ? instead of a Y the sending
failed, but asterisk doesn't retry. Next to this, no error in the logs.
Am I doing something wrong, or is there an additional app that can fix
this/monitors logs and does retries or so?
Thanks in advance.
Hi,
This won't answer your question, sorry.
How are you sending SMS ?
I'm in NL too, and can't seem to find a way to send SMS with
asterisk. The only way I found was some service on the
internet that sells SMS credits for asterisk users but it
would be nice to know how you are doing it.
Well, I found the specs on the KPN Site:
http://www.kpn.com/kpn/show/id=362543/sc=656f2e
And basicly the command for smsq is:
/usr/sbin/smsq --motx-channel=Zap/g1/067364 Number Message
It also works with chan_capi btw. Receiving is done by:
[pstn-inbound]
; SMS
exten = ${EDN_FAX}/06736400,1,Goto(s-sms-inbound,${EXTEN},1)
exten = ${EDN_WORK}/06736400,1,Goto(s-sms-inbound,${EXTEN},1)
exten = ${EDN_DIAL}/06736400,1,Goto(s-sms-inbound,${EXTEN},1)
exten = ${EDN_MAIN}/06736400,1,Goto(s-sms-inbound,${EXTEN},1)
Then s-sms-inbound contains:
[s-sms-inbound]
exten = _X.,1,NoOp(Receiving SMS from ${CALLERIDNUM})
exten = _X.,2,Answer
exten = _X.,3,Wait(1)
exten = _X.,4,SMS(incoming|a)
exten = _X.,5,System(/usr/local/AstSMSHandler/SMS2Email.pl -q)
exten = _X.,6,Hangup
And works well, btw only as mentioned the sending of the sms sometimes
failes.
--
Met Vriendelijke groet/Yours Sincerely
Stijn Jonker [EMAIL PROTECTED]
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Re: [Asterisk-Users] iax.cc opinion request
On 10/07/05, trixter http://www.0xdecafbad.com [EMAIL PROTECTED] wrote: I am considering using iax.cc (sixtel) and wondering if anyone had opinions, good or bad. Are there outages with any regularity? How responsive are tech support? How is packet loss? I am particularly interested in termination to the UK, but will accept any comments people have. For UK termination try gradwell (gradwell.com) or magrathea telecom (magrathea-telecom.co.uk), also check sipgate (sipgate.co.uk) May not be the cheapest around but quality costs money (as we all know). -Subhi ___ 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] VM Outcall: Rube Goldberg Edition
Resent to the list since I didn't think you would mind.
Kevin wrote:
Eric,
I have been using your vm outcall script for some time and it has worked
well. Thanks for your efforts.
I am trying to re-install and I can't seem to get a call file generated.
I have set up postfix and in the log it appears that it pipes the
message to the vmoutcall script. But that's as far as I get.
I was hopeful that you could offer a suggestion on how to debug the
activity of the script so I can try to figure out why it isn't working.
I don't provide support for any of the sample scripts, etc that I write.
The original VM Outcall was very primitive. Now that Asterisk has the
externnotify= option I have a much better, but still pretty primitive,
script.
In /etc/voicemail.conf, in [general] put:
===
externnotify=/usr/local/bin/vm-notify.pl
===
Here is the vm-notify.pl script:
===
#!/usr/bin/perl -w
use Fcntl;
use Fcntl :flock;
$dial_context=local-access;
($vm_box, $vm_context) = $ARGV[1] =~/(.*)\@(.*)/;
$current_vm_context = ;
if(!sysopen($vm_conf_file_handle, /etc/asterisk/voicemail.conf, O_RDONLY))
{
printf(Cannot open /etc/asterisk/voicemail.conf!\n);
exit(1);
}
while($vm_conf_line = $vm_conf_file_handle) {
chomp($vm_conf_line);
if((substr($vm_conf_line,0,1) eq ;) || (length($vm_conf_line) == 0)) {
next;
}
($tmp_vm_context) = $vm_conf_line =~ /\[(.*)\]/;
if(defined($tmp_vm_context)) {
if($current_vm_context ne ) {
exit(0);
}
if($tmp_vm_context eq $vm_context) {
$current_vm_context = $vm_context;
next;
}
} else {
if($current_vm_context eq $vm_context) {
($tmp_vm_box) = $vm_conf_line =~ /(\d+)/;
if($tmp_vm_box eq $vm_box) {
($dial_dest) = $vm_conf_line =~ /.*notify=(\d+)/;
if(!defined($dial_dest)) {
exit(0);
}
close($vm_conf_file_handle);
# If there's already a .call file for this mailbox then don't
do anything.
# If there isn't already a .call file then create it.
#$call_file_name = /tmp/ . $vm_box . .call;
$call_file_name = /var/spool/asterisk/outgoing/ . $vm_box .
.call;
if(!sysopen($call_file_handle, $call_file_name,
O_WRONLY|O_CREAT|O_EXCL)) {
exit(0);
}
flock($call_file_handle, LOCK_EX);
# Set the access and modification times to be 10 years in the
future so
# Asterisk will ignore this file while we are doing stuff with it.
$long_time = time() + (10 * 365 * 24 * 60 * 60);
utime($long_time, $long_time, $call_file_name);
srand;
$call_delay=300 + rand(120);
# Build our .call file.
printf($call_file_handle Channel:
Local/[EMAIL PROTECTED], $vm_box, $vm_context, $dial_dest,
$dial_context);
printf($call_file_handle WaitTime: 30\n);
printf($call_file_handle RetryTime: %i\n, 60 + rand(5));
printf($call_file_handle MaxRetries: 12\n);
printf($call_file_handle Context: vm-notify\n);
printf($call_file_handle Extension: s\n);
printf($call_file_handle Priority: 1\n);
printf($call_file_handle Callerid: Voicemail Notify
\9852463509\\n);
printf($call_file_handle SetVar: VM_BOX=%s\n, $vm_box);
printf($call_file_handle SetVar: VM_CONTEXT=%s\n, $vm_context);
# Unlock and close the file.
flock($call_file_handle, LOCK_UN);
close($call_file_handle);
# Set the access and modification times to be 10 mins in the
future so
#Asterisk will delay for 10 mins before processing this .call file
$short_time = time() + $call_delay;
utime($short_time, $short_time, $call_file_name);
exit(0);
}
} else {
next;
}
}
}
===
In /etc/asterisk/extensions.conf you need to put this:
===
[vm-notify]
exten = _^X.,1,Cut(VM_BOX=EXTEN,^,2)
exten = _^X.,2,Cut(VM_CONTEXT=EXTEN,^,3)
exten = _^X.,3,Cut(DIAL_DEST=EXTEN,^,4)
exten = _^X.,4,Cut(DIAL_CONTEXT=EXTEN,^,5)
exten = _^X.,5,HasNewVoiceMail([EMAIL PROTECTED])
exten = _^X.,6,Answer
exten = _^X.,7,Hangup
exten = _^X.,106,Dial(Zap/g2/${DIAL_DEST},20)
exten = _^X.,107,Noop(DIALSTATUS=${DIALSTATUS})
exten = _^X.,108,GotoIf($[${DIALSTATUS} = CHANUNAVAIL]?111)
exten = _^X.,109,GotoIf($[${DIALSTATUS} = CONGESTION]?111)
exten = _^X.,110,Answer
exten = _^X.,111,Hangup
exten = s,1,Wait(1)
exten = s,2,HasNewVoiceMail([EMAIL PROTECTED],VM_NUMBER)
exten = s,3,Hangup
exten = s,103,SetVar(LOOP=1)
exten = s,104,ResponseTimeout(1)
exten = s,105,Background(vm-youhave)
exten = s,106,SayNumber(${VM_NUMBER})
exten = s,107,Background(vm-INBOX)
exten
[Asterisk-Users] SIPGetHeaders for chan_sip (derived from chan_sip2)
hello how to drive SIPGetHeaders from chan_sip2 as described in http://www.voip-info.org/tiki-index.php?page=PortaOne+Radius+auth thanks, Kamran Ahmad __ Discover Yahoo! Find restaurants, movies, travel and more fun for the weekend. Check it out! http://discover.yahoo.com/weekend.html ___ 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.cc opinion request
On Sun, 2005-07-10 at 22:06 +0100, Subhi S Hashwa wrote: On 10/07/05, trixter http://www.0xdecafbad.com [EMAIL PROTECTED] wrote: I am considering using iax.cc (sixtel) and wondering if anyone had opinions, good or bad. Are there outages with any regularity? How responsive are tech support? How is packet loss? I am particularly interested in termination to the UK, but will accept any comments people have. For UK termination try gradwell (gradwell.com) or magrathea telecom (magrathea-telecom.co.uk), also check sipgate (sipgate.co.uk) May not be the cheapest around but quality costs money (as we all know). When I said termination I think the perspective is backwards. This is the 2nd email that seemed to come to the same comclusion. I want to call a UK number where they will terminate on the UK PSTN. I will look at those providers and see about their rates (I have already looked at sipgate though). -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ 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] sound files
where should the sound (.gsm) files be located? Currently the are in /usr/src/asterisk/sounds. I feel they should be located else ware, like in /etc/asterisk/sounds, I've copied a file into this directory but still no luck. What am I missing? Thanks Chad ___ 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] sound files
Chadwick E. Labno wrote: where should the sound (.gsm) files be located? Currently the are in /usr/src/asterisk/sounds. I feel they should be located else ware, like in /etc/asterisk/sounds, I've copied a file into this directory but still no luck. What am I missing? Thanks Chad On my system, RH9 running a version of HEAD from late Feb, the sound files are located in /var/lib/asterisk/sounds. Can't say about other distros and other versions. John Novack ___ 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] IAX2 softphone for Pocket PC
On Sun, 10 Jul 2005 16:24:19 +0200, Androtech wrote: Does anybody know an IAX2 Softphone for Pocket PC? Ciao Andro I wish . . ___ 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] DELL 2800 : PCI Parity error
I too had this problem, on a 2850, as well as the occasional missed IRQ. I went through all the usual zaptel tuning stuff Disabled fb, disabled ht, disabled acpi (left io-apic enabled), then moved irq affinity of zaptel card to second CPU so all interrupts from zaptel are on their own. My systems now run close to 100% in zttest, never miss an irq and don't seem to generate PCI parity errors any more. I don't know if I've fixed it, but you should really go through the whole process anyway. == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 FWD: 512237 ICQ: 5662270 == list wrote: Still not resolved On Wed, 2005-06-08 at 01:16, David John Walsh wrote: Frank Did you ever resolve this? If so what was the issue? On 03/05/05, list [EMAIL PROTECTED] wrote: Hi, I am struggling to get rid of a conflict on DELL 2800 : PCI Parity error (EB113 on the display) I am learning linux and asterisk as I go along, there might be obvious things I should know, but bear with me. From demsg below my 2 digium cards installed are listed (no config or connections done to digium cards yet), the conflict is with the TDM400P card, without that card, in any slot, no alarm. Zapata Telephony Interface Registered on major 196 Registered Tormenta2 PCI Controller version: 24 FALC version: TE110P: Setting up global serial parameters for E1 FALC V1.2 TE110P: Successfully initialized serial bus for card Found a Wildcard: Digium Wildcard TE110P T1/E1 Freshmaker version: 71 Freshmaker passed register test Uhhuh. NMI received. Dazed and confused, but trying to continue You probably have a hardware problem with your RAM chips Module 0: Installed -- AUTO FXS/DPO Module 1: Not installed Module 2: Not installed Module 3: Installed -- AUTO FXO (FCC mode) Found a Wildcard TDM: Wildcard TDM400P REV E/F (4 modules) Registered tone zone 8 (Norway) TE110P: Span configured for CCS/HDB3/CRC4 Calling startup (flags is 4099) wcte1xxp: Setting yellow alarm usb.c: registered new driver wcusb Wildcard USB FXS Interface driver registered TE110P: Span configured for CCS/HDB3/CRC4 Calling startup (flags is 4099) Registered tone zone 8 (Norway) TE110P: Span configured for CCS/HDB3/CRC4 Calling startup (flags is 4099) Registered tone zone 8 (Norway) ramchip problem is false, without the card all ok, ramtests on machine as well. lsmod shows wcusb driver on zaptel, I dont need that, can I remove it? is that a problem or not? # lsmod Module Size Used byNot tainted usbserial 23964 0 (autoclean) (unused) lp 9156 0 (autoclean) parport38848 0 (autoclean) [lp] autofs416984 0 (autoclean) (unused) wcusb 19552 0 (unused) wctdm 41088 0 (unused) wcte11xp 22048 0 (unused) zaptel182080 4 [wcusb wctdm wcte11xp] e1000 77884 1 (autoclean) floppy 57552 0 (autoclean) sg 37388 0 (autoclean) microcode 6912 0 (autoclean) ide-cd 34016 0 (autoclean) cdrom 32896 0 (autoclean) [ide-cd] keybdev 2976 0 (unused) mousedev5688 1 hid22308 0 (unused) input 6176 0 [keybdev mousedev hid] ehci-hcd 20776 0 (unused) usb-uhci 26860 0 (unused) usbcore81152 1 [usbserial wcusb hid ehci-hcd usb-uhci] ext3 89960 6 jbd55060 6 [ext3] megaraid2 38344 7 diskdumplib 5228 0 [megaraid2] sd_mod 13904 14 scsi_mod 115112 2 [sg megaraid2 sd_mod] finally my interrupts, bit confusing to me, looks like I have dual processor, can see the NMI but what else can be found here? # cat /proc/interrupts CPU0 CPU1 0:32983953303167IO-APIC-edge timer 1: 3300 2876IO-APIC-edge keyboard 2: 0 0 XT-PIC cascade 8: 0 1IO-APIC-edge rtc 12: 236637 237965IO-APIC-edge PS/2 Mouse 14: 261779 262965IO-APIC-edge ide0 16: 0 0 IO-APIC-level usb-uhci 18: 0 0
Re: [Asterisk-Users] sound files
If you do a make install samples in the asterisk src dir, it will put them into /var/lib/asterisk/sounds Chadwick E. Labno wrote: where should the sound (.gsm) files be located? Currently the are in /usr/src/asterisk/sounds. I feel they should be located else ware, like in /etc/asterisk/sounds, I've copied a file into this directory but still no luck. What am I missing? Thanks Chad ___ 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] sound files
Right on the money, thanks quick replies. Chad Chadwick E. Labno wrote: where should the sound (.gsm) files be located? Currently the are in /usr/src/asterisk/sounds. I feel they should be located else ware, like in /etc/asterisk/sounds, I've copied a file into this directory but still no luck. What am I missing? Thanks Chad ___ 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] phantom incomming calls from asterisk
About once a day I have noticed a phantom incoming call with a caller ID of [EMAIL PROTECTED]cut off. When I answer the call there is a dial tone and the call is disconnected. Any clues? David Koski David and List, I am having the same problem. I have an * box at my house with 1 zap (pstn on a X100p clone from digit networks) channel and one sip(linksys ATA). I am getting ring on the ATA but there is no call comming in from the pstn. The following is the CLI output when this happens. I know that there is no call on the pstn because i have an emergency phone(frequent power outages) still connected to the PSTN parallel to the * box and it never rings. All the SIP stuff is on an internal lan only. I only call out on PSTN since all I have available here in nowheare land is dial up :-( All work flawlessly except for this one problem. - Starting simple switch on 'Zap/1-1' Jul 8 13:49:23 NOTICE[6150]: chan_zap.c:5405 ss_thread: Got event 2 (Ring/Answered)... -- Executing Dial(Zap/1-1, sip/677|35) in new stack -- Called 677 -- SIP/677-55a8 is ringing == Spawn extension (default, s, 1) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' Is there anything for Zap like sip debug? My first guess is that I am getting some sort of blip in ring voltage on the PSTN but have no way to prove this. As a posible logic check I unplugged from PSTN, which put zap into Red alarm of course, and then i get no phantom calls. Is there something in the zap driver that shuts down when in red alarm? Any Ideas? Try this in zapata.conf for fun: busydetect=yes busycount=6 Let us know if it makes a difference. I have added this to zapata.conf. Will let you know what happens. can you tell me why this might help or point me to a wiki/google? Going from memory only (which might be less then accurate), the busycount parameter essentially extends zap detect time. The comments in zapata.conf refer to detecting busy tone, but something from past memory says the parameter affects more then just busy tone detection. The default value is 4 but I've been using 6 or at least a year with an x100p followed by a TDM04b, and I don't have the false ring issues. Sure wish I would have kept a diary of config changes over the last two years rather then rely on memory. It would have been helpful more than once. :( Well I added these settings to zapata.conf and am still getting the phantom rings, 2 so far this morning! have been watching ztmonitor and am seeing that that rx audio level is showing a constant ###* with an rx gain setting of -7.5 in zapata.conf. If i set gain much less it gets hard to hear voice from callers. With gain at 0.0 i get * with some peaks above this. Is this normal? John M After watching ztmonitor i have found that the rx audio goes full scale when i get the phantom rings, same as with an actual call. I think I am proving to myself that the problem is in the pots line? I am going to try and put a meter on the pots line and see if I am getting ring voltage on the phantom calls. John M ___ 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] [EMAIL PROTECTED] and Cisco 7910
Hi,
Thanks for your help. Changed everything that you said but the same problem stills remain. The phone log in but can not placer nor recive calls.
I have also changed the context to context = sccp.
I have uploaded the files I have changed. Please remember that I am using [EMAIL PROTECTED] not plain asterisk.
The changedfiles can be downloaded from:
http://www.amsystems.cc/7910/sccp.zip
Gracias,
Javier
P.S. My broadvoice pwd has been changed for security.
[EMAIL PROTECTED] wrote:
Javier Chia <[EMAIL PROTECTED]>: I have uploaded the all the .conf files and screenshots of the log and Xlite.well let's start from extensions.conf; Cisco 7910replace [121]with[sccp]because in your sccp.conf the context is sccpexten = 121,1,SetCalledParty("PRUEBA"121)exten = 121,2,Dial(PRUEBA/Test1,10,tr)the dial cmd is wrong, this is the correct one (according to your sccp.conf):exten = 121,2,Dial(SCCP/ian,10,tr)exten = 121,3,Voicemail,u121exten = 121,102,Voicemail,b121syntax errors on sccp.confreplace[SEP0008E399E223] ]with[SEP0008E399E223]callwaiting = 1 is deprecated, useincominglimit = 1intercoms are not implemented so you can remove these lines.[intercom]description = Reception Intercomdevic
e =
SEP0008E399E223; device = SEP000AB7567E18dígame si todo trabaja :-)Sergio___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users
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RE: [Asterisk-Users] Uniden UIP 200 and Asterisk.
I've taken both your advice, and the advice from carlos. I changed the sip.conf file I had to have a [heath] entry that matched my unidenMAC.txt phone's config... (Actualy once realizing how this related I changed it to the default dial number for the phone...) So now my sip.conf looks like this: [31521] Username = heath Secret = heath Type = friend Qualify = 600 Defaultip = 172.28.184.105 Context = sip Nat = no My phones config (unidenMAC.txt) looks like this: # Sip Settings MyLcdDisplay 31521 MyDialNumber 31521 DisplayName heath UserNameForProxy heath PasswordForProxy heath UserNameForRegistrar heath PasswordForRegistrar heath When I start * with 3v's I get this error after connecting the phone: *CLI Jul 10 20:17:42 NOTICE[31582]: chan_sip.c:7733 handle_request: Registration from 'sip:[EMAIL PROTECTED]' failed for '172.28.184.105' This error repeats. 172.28.184.28 is my * box. I've dinked around with the settings, tried taking out defaultip = and used host = dynamic booting the phone with dhcp enabled. I still simply can't get the phone to register with *. I've upgraded to latest firmware. I know this is probably something kind of stupid and for that I apologize, but again, another round of advice would be tremendously appreciated. Thanks, heath -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Carlos Alperin Sent: Tuesday, July 05, 2005 3:53 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Uniden UIP 200 and Asterisk. Importance: High Ok, The message is quite elocuent. The phones didn't register at all. Put your qualify=600 to avoid timeouts by now. Take off the dtmf right now, And change your section on the sip.conf to heath. (I don't see any association on the unidenMAC.txt to uip200 section) Carlos Alperin Senior System Engineer Seneca Communications, LLC [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Heath Oderman Sent: Tuesday, July 05, 2005 2:06 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Uniden UIP 200 and Asterisk. Hi, I'm new to asterisk, and have a uniden UIP 200 that I got off of ebay. I'm having trouble getting the phone to register with asterisk. I've tried a few different settings. I'd be extremely grateful if someone with a similar setting could give me the sip.conf block for the UIP and the settings you're using in uniden.txt. Here's what I have currently: IP of phone is 172.28.184.105 In sip.conf - [uip200] username = heath secret = happy type = friend qualify = no host = dynamic defaultip = 172.28.184.105 dtmfmode = rfc2833 context = sip nat=no In unidenMAC.txt - # Sip Settings MyLcdDisplay 31521 MyDialNumber 703XXX DisplayName 31521 UserNameForProxy heath PasswordForProxy happy UserNameForRegistrar heath PasswordForRegistrar happy The output from asterisk is, of course: *CLI Jul 4 15:33:15 NOTICE[22905]: chan_sip.c:7733 handle_request: Registration from 'sip:[EMAIL PROTECTED]' failed for '172.28.184.105' Jul 4 15:33:45 NOTICE[22905]: chan_sip.c:7733 handle_request: Registration from 'sip:[EMAIL PROTECTED]' failed for '172.28.184.105' Jul 4 15:34:15 NOTICE[22905]: chan_sip.c:7733 handle_request: Registration from 'sip:[EMAIL PROTECTED]' failed for '172.28.184.105' I've tried a few variants without much luck. Any pointers would be greatly appreciated. Thanks in advance for any help you can offer. heath Transparent Logic Technologies Heath Oderman 757-410-2593 x 113 [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 ___ 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] NoOp
I believed from reading that NoOp would display something on the
console. I assume the console is * in the foreground. During testing,
I've often been running * as:
asterisk -C/etc/asterisk.inX/asterisk.conf -cvnf
Does that qualify as a console? Does asterisk -r qualify as a
console?
Because nothing from any NoOp has ever shown up there, or anywhere else
I can find (from extensions.conf):
exten = s,1,NoOp,internal dial
exten = s,2,NoOp,${ARG1}
exten = s,3,NoOp,${ARG2}
exten = ${CO1CID},2,NoOp,${CALLERID}
I'm using 1.0.9 on Gentoo.
I also tried:
exten = s,1,NoOp(internal dial)
exten = s,2,NoOp(${ARG1})
exten = s,3,NoOp(${ARG2})
exten = ${CO1CID},2,NoOp(${CALLERID})
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RE: [Asterisk-Users] IAXphone - ip address - extension number.
Ok, It will sound stupid, but then the process is zlib first, and asterisk after, and that is all? Regards, Carlos -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: Sunday, July 10, 2005 12:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] IAXphone - ip address - extension number. If you don't mind, can you explain me a little more the ztcfg problem. My experience with that is null, but I need to setup a box for testing purposes only, and I don't want to but a TDM card only for make it work. I know people that has asterisk running on SIP that don't use zaptel Zapata just they do sip also to their provider (which is not my my case) but I never did that. So I plan to install it with Ztdummy on top of FC2. FWIW, I've been running asterisk on a RHv9 laptop (for client demo purposes), connecting it via iax to our office system, and never worried about ztdummy, etc. Obviously, the laptop has no zap cards. This demo never includes meetme, etc. I also have a backup system (to our office system) running RHv9, and it connects and functions just fine to the primary office system via iax. Sip phones work fine on this backup system as well. That backup system was just recently replaced with an FC3 system, and I have no doubt whatsoever it will function just fine without zap cards although that system has not yet been configured or tested. So, don't be too concerned with making ztdummy (etc) function unless you truly want to support those asterisk apps that need it (eg, meetme and whatever else the wiki points out). ___ 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] SIP phone w/ XML browser
I use the polycom sip 500's with *. They are great. It also has a services buttons for XML services. I haven't looked in to using it just yet. Peace out, BrianOn 7/9/05, Mike Clark [EMAIL PROTECTED] wrote: Pavel Jezek wrote: thank you Brian, but seems, that Polycom phones are not very good option for general corporate use and even not for use with asterisk (* explicitly unsupported!), look: from voipsupply.com: Please Note: Polycom phones are not supported under Asterisk Open Source PBX. from Polycom FAQ: Can the phones be used independent of a Technology Partner's platform? No. In order to support full business phone features, the SoundPoint IP is required to operate in conjunction with Partners' IP PBX... Can the phones support LDAP directories? Currently there is no support for directories like LDAP. Is there a web browser built into the phone? Polycom does not currently support this capability. I found avaya phones, that have nice features as I mentioned before (e.g. xml browser) , any experiance with avaya SIP phones and their cost? PJWell, we have over 100 Polycom phones deployed with Asterisk in acorporate environment and they are working extremely well.___Asterisk-Users mailing list Asterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- --Brian McManus --(801) 652-5667 ___ 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] DELL 2800 : PCI Parity error
I had the same problem with a Dell PowerEdge 800 server and a TDM400 card. I talked to Digium and they suggested a workaround by adding a NMI flag reset in the Linux boot file. This only prevents a system lockup. The system worked fine even with the blinking orange light and the dazed and confused comment from the modprobe command. I have heard that the new firmware on the TDM400P card has fixed this problem, but have not experienced that first hand. In the same machine I am using a new TE110P with no problems at all. Syed Akbar Alico Systems Inc www.alicosystems.com Tel: 562-436-1510 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rod Bacon Sent: Sunday, July 10, 2005 3:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] DELL 2800 : PCI Parity error I too had this problem, on a 2850, as well as the occasional missed IRQ. I went through all the usual zaptel tuning stuff Disabled fb, disabled ht, disabled acpi (left io-apic enabled), then moved irq affinity of zaptel card to second CPU so all interrupts from zaptel are on their own. My systems now run close to 100% in zttest, never miss an irq and don't seem to generate PCI parity errors any more. I don't know if I've fixed it, but you should really go through the whole process anyway. == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 FWD: 512237 ICQ: 5662270 == list wrote: Still not resolved On Wed, 2005-06-08 at 01:16, David John Walsh wrote: Frank Did you ever resolve this? If so what was the issue? On 03/05/05, list [EMAIL PROTECTED] wrote: Hi, I am struggling to get rid of a conflict on DELL 2800 : PCI Parity error (EB113 on the display) I am learning linux and asterisk as I go along, there might be obvious things I should know, but bear with me. From demsg below my 2 digium cards installed are listed (no config or connections done to digium cards yet), the conflict is with the TDM400P card, without that card, in any slot, no alarm. Zapata Telephony Interface Registered on major 196 Registered Tormenta2 PCI Controller version: 24 FALC version: TE110P: Setting up global serial parameters for E1 FALC V1.2 TE110P: Successfully initialized serial bus for card Found a Wildcard: Digium Wildcard TE110P T1/E1 Freshmaker version: 71 Freshmaker passed register test Uhhuh. NMI received. Dazed and confused, but trying to continue You probably have a hardware problem with your RAM chips Module 0: Installed -- AUTO FXS/DPO Module 1: Not installed Module 2: Not installed Module 3: Installed -- AUTO FXO (FCC mode) Found a Wildcard TDM: Wildcard TDM400P REV E/F (4 modules) Registered tone zone 8 (Norway) TE110P: Span configured for CCS/HDB3/CRC4 Calling startup (flags is 4099) wcte1xxp: Setting yellow alarm usb.c: registered new driver wcusb Wildcard USB FXS Interface driver registered TE110P: Span configured for CCS/HDB3/CRC4 Calling startup (flags is 4099) Registered tone zone 8 (Norway) TE110P: Span configured for CCS/HDB3/CRC4 Calling startup (flags is 4099) Registered tone zone 8 (Norway) ramchip problem is false, without the card all ok, ramtests on machine as well. lsmod shows wcusb driver on zaptel, I dont need that, can I remove it? is that a problem or not? # lsmod Module Size Used byNot tainted usbserial 23964 0 (autoclean) (unused) lp 9156 0 (autoclean) parport38848 0 (autoclean) [lp] autofs416984 0 (autoclean) (unused) wcusb 19552 0 (unused) wctdm 41088 0 (unused) wcte11xp 22048 0 (unused) zaptel182080 4 [wcusb wctdm wcte11xp] e1000 77884 1 (autoclean) floppy 57552 0 (autoclean) sg 37388 0 (autoclean) microcode 6912 0 (autoclean) ide-cd 34016 0 (autoclean) cdrom 32896 0 (autoclean) [ide-cd] keybdev 2976 0 (unused) mousedev5688 1 hid22308 0 (unused) input 6176 0 [keybdev mousedev hid] ehci-hcd 20776 0 (unused) usb-uhci 26860 0 (unused) usbcore81152 1 [usbserial wcusb hid ehci-hcd usb-uhci] ext3
Re: [Asterisk-Users] NoOp
Maybe it shows up after a certain verbosity level. Try asterisk -r
When I do that NoOps always show up.
MARK.
George Garvey wrote:
I believed from reading that NoOp would display something on the
console. I assume the console is * in the foreground. During testing,
I've often been running * as:
asterisk -C/etc/asterisk.inX/asterisk.conf -cvnf
Does that qualify as a console? Does asterisk -r qualify as a
console?
Because nothing from any NoOp has ever shown up there, or anywhere else
I can find (from extensions.conf):
exten = s,1,NoOp,internal dial
exten = s,2,NoOp,${ARG1}
exten = s,3,NoOp,${ARG2}
exten = ${CO1CID},2,NoOp,${CALLERID}
I'm using 1.0.9 on Gentoo.
I also tried:
exten = s,1,NoOp(internal dial)
exten = s,2,NoOp(${ARG1})
exten = s,3,NoOp(${ARG2})
exten = ${CO1CID},2,NoOp(${CALLERID})
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Re: [Asterisk-Users] iax.cc opinion request
trixter http://www.0xdecafbad.com wrote: I am considering using iax.cc (sixtel) and wondering if anyone had opinions, good or bad. Are there outages with any regularity? How responsive are tech support? How is packet loss? I am particularly interested in termination to the UK, but will accept any comments people have. Thanks I give them a thumbs down. I tried for over a month to get them to fix the dids that I bought, the vanity did I ordered never arrived, and their support system is a black hole.. your complaints go in and are promptly ignored. Im in the US though. JD ___ 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] SIP phone w/ XML browser
Brian Roy wrote: On 7/9/05, Dan Perik [EMAIL PROTECTED] wrote: PJ, You should check out the Polycom 500/501/600. I'm quite sure it has all that (although I don't use all of what you listed). IIRC, the 500's browser is crippled. I think you have to go up to the 600 to get that functionality. -Brian I should have tried it on my 501 before I went and opened my mouth. Sure enough, either it doesn't work, or I'm doing something wrong. The "Services" button is there, and the docs don't say anything about it not working, but even with it configured, it doesn't do anything. Seems to a be a "dead" button. Perhaps some firmware upgrade down the road will "turn it on". Looking through the archives I saw someone report that it did work on the 600, though. - 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] IAXphone - ip address - extension number.
If you don't mind, can you explain me a little more the ztcfg problem. My experience with that is null, but I need to setup a box for testing purposes only, and I don't want to but a TDM card only for make it work. I know people that has asterisk running on SIP that don't use zaptel Zapata just they do sip also to their provider (which is not my my case) but I never did that. So I plan to install it with Ztdummy on top of FC2. FWIW, I've been running asterisk on a RHv9 laptop (for client demo purposes), connecting it via iax to our office system, and never worried about ztdummy, etc. Obviously, the laptop has no zap cards. This demo never includes meetme, etc. I'm not actually surprised. In the Makefile you'll see somewhere around line 61 that ztdummy is commented out and, if you *want* ztdummy, you should remove the # before running make linux26. However, if you follow through the use of the variable names, you will see that ztdummy is suffiex anyway so whether you comment it in or out, ztdummy gets compiled. It would me interesting to do an lsmod | grep z on your laptop to see if zaptel ztdummy are loaded. I also have a backup system (to our office system) running RHv9, and it connects and functions just fine to the primary office system via iax. Sip phones work fine on this backup system as well. That backup system was just recently replaced with an FC3 system, and I have no doubt whatsoever it will function just fine without zap cards although that system has not yet been configured or tested. So, don't be too concerned with making ztdummy (etc) function unless you truly want to support those asterisk apps that need it (eg, meetme and whatever else the wiki points out). (I seem to remember IAX needed timing, which is why I was on the ztdummy mission - no cards in my testbox either.) Okay, just fired up the laptop and it registered with our office system just fine using iax2. The laptop is RHv9 with cvs-head from 4/25/05 with no modifications of any sort. The lsmod lists zaptel only, Used = 0, and no ztdummy listed. Hmmm - interesting. I did an rmmod on both ztdummy and zaptel and then fired up asterisk. I could still make a call from 1 extension to another, but both music_on_hold *and* IAX2 complained about timing (but I could still make the call!!). I haven't played with MOH yet, so I dont know if the sound will work or be choppy. It's been awhile, but MOH and Meetme are two apps/functions that do require zap timing. There might be other apps as well, I just don't recall which (if any) but they are likely listed on the wiki. ___ 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] phantom incomming calls from asterisk
I have had this same problem for quite some time on one of my setups. Everyday around the same time I get a phantom ring. If I'm at the location I pick up the sip phone asterisk rings and hangup. Asterisk hangs up the zap line...then answers the zap line again. This happens anywhere from 2-4 times in a row. If I am not at the location asterisk keeps the line off hook for an extended period of time, enough to where I hear a warning from my telco If you'd like to make a call please hangup and try again ( I noticed this from listening to recordings,all incoming calls are recorded ). At 3 other locations this has never happened. I previously had busydetect=yes and busycount=4. After seeing the response by Rich I changed busycount to 6. The problem remains. I'm going to try an even higher busycount number and hope things clear up. If busycound=6 didn't impact the problem, higher values won't either. I was probably wrong for suggesting busycount. ___ 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] Problems with a new box of [EMAIL PROTECTED] 1.3
Hello, I've recently installed [EMAIL PROTECTED], i'm following step by step the "new user guide" but I cannot get my X-Lite SIP phone see my [EMAIL PROTECTED] proxy... I've installed in aviertual machine (vmware) and there's some problems with the Zaptel service and I think that this is why I cannot connect. Thanks in advance. Fabrizzio Valencia ___ 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] NEWBIE Question: Asterisk with multiline/button phones
This is a very newb. question. Been using asterisk very happily now for several months and am considering getting some of those really 'cool' multi-button phones with LEDs and buttons. It's unclear to me if it is a straightforward task to actually setup a multiline 'presence' on the phones where the LED's light up when someone picks up a 'line' or is using a 'line' or puts a 'line' on hold or park and then would like to pick it up from another phone just by pushing the 'line #3' or 'line #4' button that is on hold and lit/flashing. Is this something that Asterisk actually does with ease? Or is it this a really complicated thing to accomplish setup? In particular in a sip only environment... no actual phone PSTN (pots) 'line's involved but with multiple SIP voip accounts to work like 'lines' with real PSTN phone numbers. We have several VOIP SIP accounts. Thanks take care! Steve ___ 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] Uniden UIP 200 and Asterisk.
[31521] Username = heath Secret = heath Type = friend Qualify = 600 Defaultip = 172.28.184.105 Context = sip Nat = no AFAIK, the username should match the [] at the top, eg: [heath] Username = heath Secret = heath Type = friend Qualify = 600 Defaultip = 172.28.184.105 Context = sip Nat = no and that should then register... Maybe you should also have a host = dynamic ?? Regards, Adam ___ 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.cc opinion request
trixter http://www.0xdecafbad.com wrote: I am considering using iax.cc (sixtel) and wondering if anyone had opinions, good or bad. Are there outages with any regularity? How responsive are tech support? How is packet loss? I am particularly interested in termination to the UK, but will accept any comments people have. Thanks I give them a thumbs down. I tried for over a month to get them to fix the dids that I bought, the vanity did I ordered never arrived, and their support system is a black hole.. your complaints go in and are promptly ignored. Im in the US though. JD I have to move 2 Call Centers here in Costa Rica to www.teliax.com it was caused for the poor customer service that sixtel have and they ignore ALL my troubles tickets when I lost my DIDs that I got, also we must change our toll-free numbers, because sixtel never answered to my claims to move them to teliax... :o( jat ___ 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] NoOp
On Sun, Jul 10, 2005 at 09:49:37PM -0400, MF Hulber wrote: Maybe it shows up after a certain verbosity level. Try asterisk -r When I do that NoOps always show up. Looks like you're right. Guess I never used enough v's ;) ___ 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] Problems with a new box of [EMAIL PROTECTED] 1.3
Xten can be quite a bastage to set up if yer not used to it! A lot of little gotchyas in it such as the line that says 'enabled Y/N' Defaults to N\ There a few other things in there that are tricky to say the least. Sometimes it takes me 2 times to get it to work and I've been setting em up for quite a while. I'm not familiar with [EMAIL PROTECTED] (yet) so have not seen the step-by-step If you are only using Xten (sip) to talk to asterisk or another sip phone I don't think you need zaptel. You will need zaptel if you expect to use the meetme conference app. I think you are more than likely just stuck on the xten client config itself. Give it another shot and watch the sip activity at the prompt to see what's going on when you start the Xten client. Try sip debug at the prompt. Steve Hello, I've recently installed [EMAIL PROTECTED], i'm following step by step the new user guide but I cannot get my X-Lite SIP phone see my [EMAIL PROTECTED] proxy... I've installed in a viertual machine (vmware) and there's some problems with the Zaptel service and I think that this is why I cannot connect. Thanks in advance. Fabrizzio Valencia___ 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] iax.cc opinion request
iax.cc (sixtel) is hit-or-miss. Some days they put my other providers to shame, other days I consider scuba-gear, the sound-quality is so bad. In their defense, it seems to have gotten a bit better. Customer service is non-existent. They provided a wonderful way for you to keep track of service problems by installing a trouble-ticket system. Customers seems to be the only ones using that system, however. Forget DIDs... You can request them, you will be charged, but they usually don't respond nor activate them. If you have $10 or $20 that's burning a hole in your pocket, they may be able to be your emergency backup in case your other three or four providers are out for lunch. I keep a small balance with them just in case. -Original Message- From: trixter http://www.0xdecafbad.com [mailto:[EMAIL PROTECTED] Sent: Sunday, July 10, 2005 7:10 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] iax.cc opinion request I am considering using iax.cc (sixtel) and wondering if anyone had opinions, good or bad. Are there outages with any regularity? How responsive are tech support? How is packet loss? I am particularly interested in termination to the UK, but will accept any comments people have. Thanks -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 ___ 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] Problems with a new box of [EMAIL PROTECTED] 1.3
Hi Welcome to Asterisk On Sun, Jul 10, 2005 at 09:45:19PM -0500, Fabrizzio Valencia wrote: Hello, I've recently installed [EMAIL PROTECTED], i'm following step by step the new user guide but I cannot get my X-Lite SIP phone see my [EMAIL PROTECTED] proxy... I've installed in a viertual machine (vmware) and there's some problems with the Zaptel service and I think that this is why I cannot connect. Here's another guide to follow: http://www.catb.org/~esr/faqs/smart-questions.html Specifically: http://www.catb.org/~esr/faqs/smart-questions.html#beprecise Please describe what actually happens, and not what doesn't. Configuration files snippets may help (mind the passwords!) . Domps from the asterisk CLI may help as well: script logfile asterisk -vvvr [observe, run, whatever] [press ctrl-c] [press ctrl-d] A dump of that session is now in 'logfile'. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ 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] Cepstral
On 7/10/05, Jim Archer [EMAIL PROTECTED] wrote: Thanks William and John, I'll look again for that download. Comments below... --On Sunday, July 10, 2005 1:50 PM +0200 Wilson Pickett [EMAIL PROTECTED] wrote: FWIW? I bought that voice and I find it amusing, but not ready for prime time. I had it read articles from a publication and it was ludicrous. I can understand the people talking about ATT, I think I heard a demo that was very convincing. What is ATT? Is it another text to speech engine? I installed Festival a ATT Natural voices seem to be pretty good. You can hear samples here: http://www.wizzardsoftware.com/att_desktop.php . The Rich voice I think sounds the best. They are a little better than the Cepstral voices. But the Cepstral voices are vey good also. Michael ___ 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] NoOp
On Sun, Jul 10, 2005 at 05:27:06PM -0700, George Garvey wrote: I believed from reading that NoOp would display something on the console. I assume the console is * in the foreground. astterisk -r provides basically the same functionality . Generally people overuse the option '-c' and tie asterisk to a specific terminal. What should asterisk do when that terminal will blow away? During testing, I've often been running * as: asterisk -C/etc/asterisk.inX/asterisk.conf -cvnf Does that qualify as a console? Does asterisk -r qualify as a console? Asterisk -r connects to a unix-domain-socket , whose path can be controlled in asterisk.conf (it resides in the astrundir) . If you use several config files for running several copies of asterisk, you need to provide different astrundir-s. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ 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] Problems with a new box of [EMAIL PROTECTED] 1.3
Are you getting any messages from the CLI on * pertaining to a sip user not registering? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Fabrizzio ValenciaSent: Sunday, July 10, 2005 7:45 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] Problems with a new box of [EMAIL PROTECTED] 1.3 Hello, I've recently installed [EMAIL PROTECTED], i'm following step by step the "new user guide" but I cannot get my X-Lite SIP phone see my [EMAIL PROTECTED] proxy... I've installed in aviertual machine (vmware) and there's some problems with the Zaptel service and I think that this is why I cannot connect. Thanks in advance. Fabrizzio Valencia ___ 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 download chan_sip2
hello http://www.voip-info.org/tiki-index.php?page=Asterisk+SIP+chan_sip2 where can i download chan_sip2.c thanks Kamran __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Howto get streaming mp3 at an extension?
I would simply like to dial an extension and get an individual Live MP3 stream but am unsure of how to do this. I'd like it to be different from my music on hold (not the same source) This trick works for music on hold: in musiconhold.conf ;default = mp3:/var/lib/asterisk/live,http://sourceofstream.com:8001/ I still wish to use local files for music on hold but want to dial an extension to listen to a live stream. Any ideas? Thanks! Steve ___ 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 Crashes after update
You might want to recompile the res_config_mysql or configure res_config_odbc which works via myodbc and is just as good! /b --- Anakin: “You’re either with me, or you’re my enemy.” Obi-Wan: “Only a Sith could be an absolutist.” On Jul 7, 2005, at 2:46 PM, [EMAIL PROTECTED] wrote: After doing an update from SUSE 9.2 to 9.3 and Checking out the latest from CVS, Asterisk crashes on startup with an apparent MySQL (res_config_register) error: # asterisk -vvvgc asterisk_startup_error1.log asterisk: symbol lookup error: /usr/lib/asterisk/modules/res_config_mysql.so: un defined symbol: ast_cust_config_register The log is shown below. I've seen the posts from 1/25/05 and several more recent ones regarding this same issue or a similar one with the ast_cust_config_register being undefined, however reverting to that build of 1/24/05 does not solve the problem in my case. Is there another issue with mySQL that may cause this problem? I'm using SUSE 9.3 on an Athlon 64 with 64 bit release 2.6 of Linux. I've made sure that all the ODBC and MySQL modules for SUSE 9.3 are installed. I'm a rank noob with * and would appreciate any help. Thanks!!! Log Pasted below for more info: [0;37;40m [1;30;40m == [0;37;40mParsing '/etc/asterisk/asterisk.conf': Found [1;30;40m == [0;37;40mParsing '/etc/asterisk/extconfig.conf': Found [1;30;40m == [0;37;40mParsing '/etc/asterisk/asterisk.conf': Found Asterisk CVS-HEAD, Copyright (C) 1999 - 2005 Digium. Written by Mark Spencer [EMAIL PROTECTED] == === [1;30;40m == [0;37;40mParsing '/etc/asterisk/logger.conf': Found Asterisk Event Logger Started /var/log/asterisk/event_log Asterisk Dynamic Loader loading preload modules: [1;30;40m == [0;37;40mParsing '/etc/asterisk/modules.conf': Found [1;30;40m == [0;37;40mManager registered action Ping [1;30;40m == [0;37;40mManager registered action Events [1;30;40m == [0;37;40mManager registered action Logoff [1;30;40m == [0;37;40mManager registered action Hangup [1;30;40m == [0;37;40mManager registered action Status [1;30;40m == [0;37;40mManager registered action Setvar [1;30;40m == [0;37;40mManager registered action Getvar [1;30;40m == [0;37;40mManager registered action Redirect [1;30;40m == [0;37;40mManager registered action Originate [1;30;40m == [0;37;40mManager registered action Command [1;30;40m == [0;37;40mManager registered action ExtensionState [1;30;40m == [0;37;40mManager registered action AbsoluteTimeout [1;30;40m == [0;37;40mManager registered action MailboxStatus [1;30;40m == [0;37;40mManager registered action MailboxCount [1;30;40m == [0;37;40mManager registered action ListCommands [1;30;40m == [0;37;40mParsing '/etc/asterisk/manager.conf': Found [1;30;40m == [0;37;40mParsing '/etc/asterisk/cdr.conf': Not found (No such file or directory) Jul 6 21:32:24 [1;33;40mNOTICE[0;37;40m[8492]: [1;37;40mcdr.c[0;37;40m:[1;37;40m1162[0;37;40m [1;37;40mdo_reload[0;37;40m: CDR simple logging enabled. [1;30;40m == [0;37;40mParsing '/etc/asterisk/rtp.conf': Found [1;30;40m == [0;37;40mRTP Allocating from port range 1 - 2 Asterisk PBX Core Initializing Registering builtin applications: [1;30;40m [0;37;40m[AbsoluteTimeout] [1;30;40m == [0;37;40mRegistered application '[1;36;40mAbsoluteTimeout[0;37;40m' [1;30;40m [0;37;40m[Answer] [1;30;40m == [0;37;40mRegistered application '[1;36;40mAnswer[0;37;40m' [1;30;40m [0;37;40m[BackGround] [1;30;40m == [0;37;40mRegistered application '[1;36;40mBackGround[0;37;40m' [1;30;40m [0;37;40m[Busy] [1;30;40m == [0;37;40mRegistered application '[1;36;40mBusy[0;37;40m' [1;30;40m [0;37;40m[Congestion] [1;30;40m == [0;37;40mRegistered application '[1;36;40mCongestion[0;37;40m' [1;30;40m [0;37;40m[DigitTimeout] [1;30;40m == [0;37;40mRegistered application '[1;36;40mDigitTimeout[0;37;40m' [1;30;40m [0;37;40m[Goto] [1;30;40m == [0;37;40mRegistered application '[1;36;40mGoto[0;37;40m' [1;30;40m [0;37;40m[GotoIf] [1;30;40m == [0;37;40mRegistered application '[1;36;40mGotoIf[0;37;40m' [1;30;40m [0;37;40m[GotoIfTime] [1;30;40m == [0;37;40mRegistered application '[1;36;40mGotoIfTime[0;37;40m' [1;30;40m [0;37;40m[ExecIfTime] [1;30;40m == [0;37;40mRegistered application '[1;36;40mExecIfTime[0;37;40m' [1;30;40m [0;37;40m[Hangup] [1;30;40m == [0;37;40mRegistered application '[1;36;40mHangup[0;37;40m' [1;30;40m [0;37;40m[NoOp] [1;30;40m == [0;37;40mRegistered application '[1;36;40mNoOp[0;37;40m' [1;30;40m [0;37;40m[Prefix] [1;30;40m == [0;37;40mRegistered application '[1;36;40mPrefix[0;37;40m' [1;30;40m [0;37;40m[Progress] [1;30;40m == [0;37;40mRegistered application '[1;36;40mProgress[0;37;40m' [1;30;40m [0;37;40m[ResetCDR] [1;30;40m == [0;37;40mRegistered application '[1;36;40mResetCDR[0;37;40m' [1;30;40m [0;37;40m[ResponseTimeout]
