RE: [Asterisk-Users] chan_mISDN
Hi I am just wondering if chan_mISDN is a worthwhile alternative to zaphfc which I am having issues with. I have 2 hfc-s modem cards in my asterisk box. Any comments or advice will be appreciated. Thanks Clive On 19 Oct 2004 at 11:16, Brian West wrote: Well the error does give you some clue on whats wrong and it's done that way to give you exactly what you need to do: Use AST_DEFINE_STATIC rather than AST_MUTEXT_INITIALIZER Check out the other apps and compare them to chan_mISDN and you'll get what you need to change.. its only one line if I recall. bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Erwan DESVERGNES Sent: Tuesday, October 19, 2004 10:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] chan_mISDN Did someone have succeed to compile chan_misdn ??? Ive got an error when in try to compile chan_misdn.c:68: error: `__use_AST_MUTEX_DEFINE_STATIC_rather_than_AST_MUTEX_INITIALIZER__' undeclared here (not in a function) thanks _ Erwan Desvergnes - ANDIUM - 82/86 rue Château Gaillard 69100 Villeurbanne Tel. 04 37 43 44 45 / Fax 04 37 43 44 44 E-mail: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk not sending full 11 digits dialed....
On Tue, 19 Oct 2004, Michael Loftis wrote: We figured it out. Well I did. You pretty much have to use pridialplan=unknown in zapata.conf it looks like, with the others libpri seems to try to get stupid with the actual digits sent/coded to the remote switch. Also, your telco may interpret the digits you send differently. Unknown for TON/NPI tells the telco to interpret the digits as if they were dialed on a pots line which is usually what humans want. Peter ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] meetme latency
I am pretty sure that I had used meetme in the past (many months ago) with great results. Small number of users, mixed connections, IAX2 and SIP. For the past month or so, meetme has been a real pain due to very large latency. I can take 2 phones on the local lan and still get many seconds of latency. This makes it really hard to carry on a conversation. If I try to have folks join in over the net, we end up with 4 to 5 second latency. Is this normal, or do I have a problem. I am running 2.6.8ish kernel with no zap hardware. I am using the 2.6ish ztdummy. zttest looks ok. Echo test and phone calls are great. I think it is only when I get into the pseudo zap driver that I start having problems. Is it time for me to check out app_conference? -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
AW: [Asterisk-Users] CAPI and Asterisk (with AVM ISDN Card)
HI Matt I'm unsure about Fedore, but hav several instances of * running with different avm cards running find (usb, fritz and c2) on debian stable (woody). The tricky part is to find the correct verion of the avm drivers to load (since they are compiled for suse. Does capiinfo show anything? Have you enabled capi in the kernel? Best regards, Pascal (also from switzerland) _ Von: Mateo Meier [mailto:[EMAIL PROTECTED] Gesendet: Mittwoch, 20. Oktober 2004 00:31 An: [EMAIL PROTECTED] Betreff: [Asterisk-Users] CAPI and Asterisk (with AVM ISDN Card) Hello Guys Im trying to get Asterisk with my AVM fritz Card (ISDN) to work. ( fedora core 1 ) I did found a easy how to.. it was posted from someone here on this Mailing List Im referring to http://lists.digium.com/pipermail/asterisk-users/2004-June/052118.html MY PROBLEM: I can't get CAPI to work ;-) The how to (http://lists.digium.com/pipermail/asterisk-users/2004-June/052118.html) is assuming you have already installed capi and are able to edit the the /etc/capi.conf File. Does anybody knows what version of capi is needed ? I tried to install a capi rpm.. but after the capi rpm installation, there seems to be no /etc/capi.conf What kind of capi version do I need ? capi4k-utils ? or just any capi rpm ? Thank you so much for your feedback Regards from Switzerland Matt ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Comments on proposed * setup
Are you able to consider IP phones? Once you take the per-port cost of analogue phones into account, IP phones may end up being less expensive. As for the trunks, it sure is difficult to imagine that one would forgo digital trunking in favour of such a large a quantity of analogue circuits. Are you sure your customer cannot be swayed on this point? That many analogue trunks is going to be a pain. (in most of North America, the monthly costs of the analogue circuits would be more expensive as well). Cheers, Jim. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of el Flynn Sent: October 19, 2004 11:39 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Comments on proposed * setup Hi all, Just wanted to see what you guys have to say about the setup we're planning to install - 16 incoming POTS lines, 50 extensions. As it is, I've got two options: 1) Lots of ATAs 1 x * server 4 x TDM04B for 16 incoming lines (can't do fractional E1 - client's requirements) 25 x Sipura SPA-2000 connected to a total of 50 analog phones 2) Channel banks 2 x * servers (due to PCI slot limitations...) 4 x TDM04B (as above) 2 x T100P 2 x 24-port channel bank 1 x Sipura SPA-2000 connected to 2 analog phones Option #1 is cheaper but (i suppose) more hassles in terms of SIP.conf, wall warts etc. Option #2 is more expensive (for me anyways) but more manageable (?) I'm leaning more towards this but am concerned about costs. Anyone can provide hints, suggestions, comments as to why I would choose one option over the other? I _can't_ do anything about costs, due to certain licensing and legal issues here in my country, so suggestions like buy 'em from eBay and save cost won't do any good :) cheers, flynn ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/ asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.775 / Virus Database: 522 - Release Date: 08/10/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.775 / Virus Database: 522 - Release Date: 08/10/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X100P red alert
On Wed, 20 Oct 2004 06:34:45 +0200, Alex van Es [EMAIL PROTECTED] wrote: I was just trying to find out if the fact that the driver doesn't load is because it's not plugged into the phoneline, thats all. Sorry but I just couldn't resist ;-) The driver should have loaded because it is the driver that detects the missing phone line and generates the RED ALARM in the first place. If it doesn't load, you should be able to see the deivce and there can't be any alarms on a device that doesn't exist. on the command line do ... ls /proc/zaptel if you don't have any directories 1, 2, 3 etc in there, then the driver didn't load. If the driver loaded, there should be a numbered directory for each FXO card or FXO/FXS module. if you then do ... ls /proc/zaptel/1 it should should show you the details of the card/module that is associated with Zap1. If it says RED ALARM, it means the card/module cannnot see any phone line. Consequently, if you try to make a phone call on that card/module, Asterisk will not be able to dial out because there is no phone line to dial out on. As a result it will give you the error message you see. If you don't have an analog phone line to test where the box is located, you could connect it with an ordinary phone wire to an FXS port on some ATA, ie a Grandstream HT286, Sipura-1/2/3K, or IAXy to mimic the phone line. rgds benjk -- Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Tokyo, Japan. NB: Spam filters in place. Messages unrelated to the * mailing lists may get trashed. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_mISDN problem
Im using avm fritz card usb with kernel 2.6 patch with mISDN. The module is load correctly when I type lsmod Ive got the following output: Module Size Used by zaptel 178308 0 avmfritz 21388 0 mISDN_isac 14336 1 avmfritz mISDN_dsp 191424 0 l3udss1 34184 0 mISDN_l2 39040 0 mISDN_l1 11016 0 mISDN_core 67168 6 avmfritz,mISDN_isac,mISDN_dsp,l3udss1,mISDN_l2,mISDN_l1 edd 9240 0 joydev 10304 0 sg 35488 0 st 39196 0 sr_mod 16292 0 ide_cd 36740 0 cdrom 36892 2 sr_mod,ide_cd nvram 8456 0 capidrv 28340 0 isdn 134092 1 capidrv slhc 7552 1 isdn capi 17728 0 capifs 5896 2 capi kernelcapi 45856 2 capidrv,capi usbserial 29040 0 parport_pc 35392 1 lp 11044 0 parport 37832 2 parport_pc,lp ohci_hcd 19204 0 speedstep_lib 3712 0 sworks_agp 9376 0 agpgart 30888 1 sworks_agp freq_table 4612 0 thermal 12680 0 processor 16552 1 thermal snd_seq_oss 31232 0 fan 4228 0 snd_pcm_oss 57512 0 snd_mixer_oss 18816 1 snd_pcm_oss button 6416 0 battery 8836 0 snd_seq_midi 9120 0 snd_seq_midi_event 7680 2 snd_seq_oss,snd_seq_midi snd_seq 54928 5 snd_seq_oss,snd_seq_midi,snd_seq_midi_event ac 4996 0 ipv6 237440 22 evdev 9728 0 snd_ens1371 23012 0 snd_rawmidi 25508 2 snd_seq_midi,snd_ens1371 snd_seq_device 8456 4 snd_seq_oss,snd_seq_midi,snd_seq,snd_rawmidi snd_pcm 97032 2 snd_pcm_oss,snd_ens1371 snd_page_alloc 11528 1 snd_pcm snd_timer 25732 2 snd_seq,snd_pcm snd_ac97_codec 62468 1 snd_ens1371 snd 61444 12 snd_seq_oss,snd_pcm_oss,snd_mixer_oss,snd_seq_midi,snd_seq_midi_event,snd_seq,snd_ens1371,snd_rawmidi,snd_seq_device,snd_pcm,snd_timer,snd_ac97_codec soundcore 8928 1 snd gameport 4736 1 snd_ens1371 usbcore 103516 4 usbserial,ohci_hcd 8139too 23168 0 mii 5248 1 8139too dm_mod 50172 0 reiserfs 241360 1 aic7xxx 177844 2 sd_mod 20096 3 scsi_mod 108748 5 sg,st,sr_mod,aic7xxx,sd_mod but when I try to start asterisk it doesnt start and Ive got the following output: [chan_misdn.so] = (Channel driver for mISDN Support (Bri/Pri)) == Parsing '/etc/asterisk/misdn.conf': Found UnLocking config_mutex == Registered channel type 'mISDN' (This driver enables the asterisk to use hardware which is supported by the ne) Locking Config Mutex UnLocking Config Mutex Init. Stack on port 1 unknown port(1) type 0x0010 init_stack: No such file or directory Please help me _ Erwan Desvergnes - ANDIUM - 82/86 rue Château Gaillard 69100 Villeurbanne Tel. 04 3743 44 45 / Fax 04 37 43 44 44 E-mail: [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Load Balaning on 2 E100P cards
Hello, I would like to know if it's possible to load balance calls on 2 E100P cards? In fact, I had an asterisk with a TE410P. 2 E1 are connected to the operator, and 2 others to an IVR PBX. Asterisk is used to place some calls in Voice over IP. I would like to know if it's possible, when I receive a call from my operator, if I can load balance it on my 2 others E1 connected to the PABX. I this case, ifone PABX fail, I still had another one. Thanks. Regards. GIBERT Frédéric Mobile: +33 (0) 6 7208 3516 Fax : +33 (0) 1 4692 0569 [EMAIL PROTECTED] http://www.viginetworks.fr Ste VIGINETWORKS 1, rue Craiova 92000 Nanterre France logo.gif___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Sipura or X100P Option
The TDM400 improves on the X100P in every way. Also, because the channels are nearly identical in the way they relate to the system, you won't have to make many changes to your dialplan to implement it. Running an external FXO interface (which will need to run over an IP link) may deliver results you weren't anticipating (echo is one that comes to mind). I'd say the TDM400 would be the thing to implement, although I would also mention that the Sipura is very well regarded product. Purchase a TDM04B bundle and you'll have your three lines, plus a fourth for future growth. You can order a TDM03B to and possibly save $80, but the sage advise is to get the hardware installed and then don't mess with it - fully provision the card when you install it and you'll never have to touch it again. Hope this helps, Regards, Jim. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brent Franks Sent: October 20, 2004 12:31 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Sipura or X100P Option Hello, Our client currently has two X100P's running in an HP box that has been running for almost a year now with no problems. They have found however that two phone lines are not enough and are bringing in a third phone line. I wouldn't expect this line to be used very often as there are only two employees in the office. I am curious which route to head. I am hesitant to throw another X100P in the box and create the potential for problems, or should I use a Sipura as an FXO device. Has anyone had any experience with Sipura as an FXO? Are there any issues I should know about? Thanks in advance, Brent D. Franks Mindworks Internet Services ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/ asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.775 / Virus Database: 522 - Release Date: 08/10/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.775 / Virus Database: 522 - Release Date: 08/10/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CAPI and Asterisk (with AVM ISDN Card)
On Wednesday 20 October 2004 00:30, Mateo Meier wrote: Does anybody knows what version of capi is needed ? try the most recent here: ftp://ftp.isdn4linux.de/pub/isdn4linux/CVS-Snapshots it did work fine for me (FC2 and debian sid) I tried to install a capi rpm.. but after the capi rpm installation, there seems to be no /etc/capi.conf cat capi.conf # card fileproto io irq mem cardnr options b1isa b1.t4 DSS10x150 7 - - P2P b1pci b1.t4 DSS1- - - - c4 /usr/sbin/c4.binDSS1- - - - c4 - DSS1- - - - c4 - DSS1- - - - P2MP c4 - DSS1- - - - P2MP c2 c2.bin DSS1- - - - c2 - DSS1- - - - t1isa t1.t4 DSS10x340 9 - 0 t1pci t1.t4 DSS1- - - - fcpci - - - - - - fcclassic - - 0x150 10 - - What kind of capi version do I need ? capi4k-utils ? or just any capi rpm ? download a tarball and install it... Maurizio ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re[2]: [Asterisk-Users] Fax over IP doesn't works
Dear Steve, SU So how does the FAX get from the fax machine to the T.38 channel SU with spandsp? In our case we will try to strip spandsp and will use directly OpenH323. We do tests for compatibility with one of the biggest national telecom and if they are OK, they will offer Asterisk based IP PBX to their clients instead Cisco. That's why we need of T.38 and G.711 fax capabilities. Also we have the problems with the following tests: 1. When Dialing of unallocated number the resposne must be Invalid Number, but the result is one of the following: Hangup, Congestion or Busy. 2. CLIP/CLIR User provided verified and passed - We can't find where we can set this bits for this services. 3. Fax T38 / g711 4. Codec negotiation: when 2 codecs are possible (G.711 and G.729), the two parties can't negotiate which codec to use. Best Regards, Miroslav Nachev Miroslav Nachev wrote: SU and exactly how does that get the FAX into the T.38 channel? :-\ Using G.711 or implementing T.38 in Asterisk or adjusting Asterisk to OpenH323 T.38. From our expirience Asterisk detect that the line is with Fax data. The problem is what next. So how does the FAX get from the fax machine to the T.38 channel with spandsp? Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] cheap gig switch? smc, netgear, or 3com?
Hello, The Smc 8508T goes for about $95, jumbo frame support, lifetime warranty but no QOS. The Netgear GS608 is $ 100, no jumbo frames, 1 year warranty, QOS, gig latency 10U max. The 3com switch reviews that I read were not happy. Does anyone hate or love their home switch? I doubt the jumbo frame support would help voip traffic, but it seems like it wouldn't hurt. I was planning on doing the QOS on linux. Gig support is wanted for file transfers and the future. Thanks to all you nice asterisk people and a few of the mean ones. Jay __ Do you Yahoo!? Yahoo! Mail Address AutoComplete - You start. We finish. http://promotions.yahoo.com/new_mail ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] meetme latency
For what it's worth, I have the same observation. Meetme used to work great, but sometime in the last few (3-4) months, it seems to have developed significant latency. Our echo test is also normal (way under a second), as are non-meetme calls. Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 On Tue, 19 Oct 2004, Bob Knight wrote: I am pretty sure that I had used meetme in the past (many months ago) with great results. Small number of users, mixed connections, IAX2 and SIP. For the past month or so, meetme has been a real pain due to very large latency. I can take 2 phones on the local lan and still get many seconds of latency. This makes it really hard to carry on a conversation. If I try to have folks join in over the net, we end up with 4 to 5 second latency. Is this normal, or do I have a problem. I am running 2.6.8ish kernel with no zap hardware. I am using the 2.6ish ztdummy. zttest looks ok. Echo test and phone calls are great. I think it is only when I get into the pseudo zap driver that I start having problems. Is it time for me to check out app_conference? -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This message has been categorized as Legitimate by Bayesian Analyzer. If you do not agree, please click on the link below to train the Analyzer. http://216.162.162.39/bt/a.aspx?M=C:%5Csmtpmail%5CBayesTraining%5C2004-10-19%5Cfdb007959f614e6190803a5c35248faeC=2 -- --- This message has been inspected by DynaComm i:mail --- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] i extension
Benjamin on Asterisk Mailing Lists wrote:
On Tue, 19 Oct 2004 17:58:00 -0400, Steve Kann [EMAIL PROTECTED] wrote:
This explicitly repeats the invalid number back to them; you could
prefix it with a message saying the number you dialed and postfix with
is invalid, blah blah..
exten = i,1,SayDigits(${INVALID_EXTEN})
exten = i,2,Goto(s,1)
It would seem that the i extension is never called unless you have
something like Background(dial-a-number) in the context.
Extensions like i, h, and the rest do not seem to be included with
include =. These extensions seem to have to be in the context they
are called from.
begin:vcard
fn:Eric Wileing
n:Wileing;Eric
email;internet:[EMAIL PROTECTED]
tel;work:504-899-1387 x2120
x-mozilla-html:FALSE
version:2.1
end:vcard
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Re: [Asterisk-Users] Asterisk on PowerPC v. Intel/AMD
On Tue, 19 Oct 2004 21:39:03 -0500, Brian McSpadden [EMAIL PROTECTED] wrote: In my case, I was running two X100P's. Not exactly the TDM40, but should be the same concept. The driver for that card is slightly different, in the fact that it uses the wcfxs kernel module (even on FXO interfaces), rather than the wcfxo module of the X100P. However, I doubt it makes a difference, the hardware should be compatible. It sure works great on my 8500. I have tested a TDM400 with 2 FXS modules on a PowerMac G3 Desktop running YDL3.0.1 and it worked fine. This was some time earlier this year, before the FXO modules were available.. rgds benjk -- Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Tokyo, Japan. NB: Spam filters in place. Messages unrelated to the * mailing lists may get trashed. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Meetme room calls quiet for some lines/callers
I just tried out the meetme room feature for the first time and found a few issues with the levels. I had three calls, one on an fxs port (TMD400), 2 on fxo ports, one fxs was a 100P the other was on the TDM400. The phone on the fxs port could hear and everyone could hear that line. The two calls on the fxo could barely hear each other. I did a little fiddling with the rx and tx gain settings in zapata.conf and this impacted the over all levels for all lines but I am not sure I completely understand what is happening here and in what direction the tx and rx are effecting things. Does anyone have some guidance on how I should change these settings or if I should be looking at something else. -- Scott Henderson == Finite Technologies Incorporated 3763 Image Drive, Anchorage, Alaska 99504 Phone: 907.337.2860, Fax: 907.333.4482 http://www.finite-tech.com http://www.chillywall.com http://www.virtuale.cc http://www.mphage.com == ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] i extension
On Wed, 20 Oct 2004 04:15:21 -0500, Eric Wieling [EMAIL PROTECTED] wrote:
Extensions like i, h, and the rest do not seem to be included with
include =. These extensions seem to have to be in the context they
are called from.
Even so, they seem to only be called from within an IVR menu where the
channel has been answered already, for example from a
Backgroun(enter-a-number) statement.
If you have this ...
[office]
;
include = local
include = national
include = international
;
exten = _20XX,1,NoOp(call for extension ${EXTEN})
exten = _20XX,2,Dial(SIP/${EXTEN},60,r)
exten = _20XX,3,Hangup
;
exten = i,1,NoOp(invalid number/extension dialled)
exten = i,2,Playback(pbx-invalid)
exten = i,3,Hangup
;
; END of this context
while all your SIP clients have context=office assigned to them in sip.conf, ...
then if somebody dials a number that doesn't match anything in the
office context nor in any included context, it will still not go to
the i extension.
It seems this is only meant for IVR menus. So, the bogons context with
the catch all _. extension seems to be the only way to catch the
misdialled numbers.
rgds
benjk
--
Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya,
Tokyo, Japan.
NB: Spam filters in place. Messages unrelated to the * mailing lists
may get trashed.
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AW: [Asterisk-Users] Follow me using a loop
Hello Drop the third line. exten = 31xxx,3,Goto(31xxx,1) 31xxx,2,Dial(IAX2/[EMAIL PROTECTED]CAPI/31xxx:079xxx,120,mgh) [context-Follow_me] exten = 31xxx,1,Wait(1) exten = 31xxx,2,Dial(IAX2/[EMAIL PROTECTED]CAPI/31xxx:079xxx,120,mgh) exten = i,1,hangup ; dialed an invalid number exten = t,1,hangup ; timeout By dropping the 3rd line the call does not reconnect if I (as callee) hangup the line, after the timeout the caller gets disconnected. What I am trying to achieve is what other vendors are calling mobile extension. Picking up the call on any device talking to the caller, hanging up the line and taking it (the same call) at the mobile phone to be able to walk away. All this works fine, even with moh for the caller while the call ist established again, but the problem is, that I (as callee) am unable to hangup the line. Which means, as long the caller stays on the line, the phones will keep ringing. I tried also to use any key (# or *) to hangup the call, but it seems not to dial the hangup extension. Maybe I'm just trying to achieve something weird. Best regards, Pascal. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sipura or X100P Option
On Wed, 20 Oct 2004 00:30:43 -0400, Brent Franks [EMAIL PROTECTED] wrote: Our client currently has two X100P's running in an HP box that has been running for almost a year now with no problems. They have found however that two phone lines are not enough and are bringing in a third phone line. I wouldn't expect this line to be used very often as there are only two employees in the office. I am curious which route to head. I am hesitant to throw another X100P in the box and create the potential for problems, It may be worth a trial. If the box has troubles with a third card, throw the toy away and get a Mac with YDL: http://www.voip-info.org/tiki-index.php?page=Asterisk+Linux+Yellow+Dog Has anyone had any experience with Sipura as an FXO? Are there any issues I should know about? It's probably OK as a backup line. We've got a problem with the SPA-3K not recognising incoming calls on a Japanese PSTN line, but that wouldn't be of concern to you if you are in the US or Canada. There is a slight Echo on the Sipura if you have to raise the TX gain, but consecutive firmware updates seem to have steadily improved this, so after a few more firmware updates, the echo might be gone entirely. Configuring the Sipura can be a bit intimidating. I have never seem any device with so many settings. However, you only need to play with a few of them, so once you worked out what to touch and what to leave alone, this will not be a big deal anymore. Somebody reported having blown up an SPA-3K and getting support can be a bit of an exercise in patience. Other than that the SPA-3K would seem like a very simple upgrade option in your case and probably good enough. rgds benjk -- Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Tokyo, Japan. NB: Spam filters in place. Messages unrelated to the * mailing lists may get trashed. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ISDN asterisk ISDN PBX possible ?
Hello May be this question is a little off topic. I like to use asterisk as follow, is that possible. NT1 ISDN from telecom((isdn 1st card ) ASTERISK (2th isdn card)) excisting regular PBX system. In this way I don't have to invest in phone hardware stuff and i can join to the low cost VOIP calls. Specialy for my tele workers. Incomming calls go transparantly through the asterisk server. Outgoing calls go through asterisk to internet VOIP provider if cheaper. I'm mostly wurry about ISDN NT1 etc. Maybe someone can point me to a allready excist lowcost device on the market. Thanks Sjaak ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP/SIMPLE, Jabber and Asterisk
Title: SIP/SIMPLE, Jabber and Asterisk Dear All, Is there an implementation of SIP/Simple for Asterisks? It would be neat to tie Asterisk to an IM like Jabber for presence. I believe this is already available for SER. Can anyone tell me if this is on the roadmap? I have been using both Asterisk and Jabber for quiet some time and would love to see these two working with each other. Would welcome any input on this. Shad Mortazavi Nexus Technical Manager n|m Nexus Management Inc Neutral Bay Sydney ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ISDN asterisk ISDN PBX possible ?
Hi Look for ISDN-Cards with an HFC-S chip and use the bristuff-tools from www.junghanns.net. The cards cost new about 30EUR each. More information about hfc-s can be found in the wiki. Regards Kai Am Mi, den 20.10.2004 schrieb sjaak imap um 11:57: Hello May be this question is a little off topic. I like to use asterisk as follow, is that possible. NT1 ISDN from telecom((isdn 1st card ) ASTERISK (2th isdn card)) excisting regular PBX system. In this way I don't have to invest in phone hardware stuff and i can join to the low cost VOIP calls. Specialy for my tele workers. Incomming calls go transparantly through the asterisk server. Outgoing calls go through asterisk to internet VOIP provider if cheaper. I'm mostly wurry about ISDN NT1 etc. Maybe someone can point me to a allready excist lowcost device on the market. Thanks Sjaak ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Kai Militzer WESTEND GmbH | Internet-Business-Provider Technik CISCO Systems Partner - Authorized Reseller Lütticher Straße 10 Tel 0241/701333-11 [EMAIL PROTECTED] D-52064 Aachen Fax 0241/911879 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] of call between ser and asterisk
Hi Guys I think my last post might have been a little long, so I'll rephrase :-) I have asterisk and ser, I just need to know the flow of the call is it xlite--ser---asterisk---xlite or xlite --ser--asterisk--ser---xlite or should it be xlite ---asterisk---ser-.xlite or xlite ---asterisk---ser---asterisk---xlite once I have this sorted, i should be able to work it out...I hope Iqbal ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] GNUGK + ASTERISK
Hello I have GNUGK already instaled and from the possible configurations, the only one that worked was this one: pwlib-1.6.6-0_11.rh9.at.src.rpm openh323-1.13.5-0_13.rh9.at.src.rpm but now I want to install the Open H.323 Channel on Asterisk and I cant, because It sais I need to have: Open H.323 v1.12.2 PWLib v1.5.2 Does Asterisk Open H.323 Channel only works with Open H.323 v1.12.2 and PWLib v1.5.2 or is there a way of putting the H.323 Channel working with pwlib-1.6.6 and openh323-1.13.5 ?? The Asterisk readme file says: if you are not using the listed versions of Open H.323 or PWlib you are on your own but this doesnt mean that it doesnt work does some of you allready putted it to work together??? Thanks Joao ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP/SIMPLE, Jabber and Asterisk
Title: Message I think I have seen such application few months agofor asterisk but it may not be open source. Anyone knows more? -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Shad MortazaviSent: 20 October 2004 11:06To: [EMAIL PROTECTED]Subject: [Asterisk-Users] SIP/SIMPLE, Jabber and Asterisk Dear All, Is there an implementation of SIP/Simple for Asterisks? It would be neat to tie Asterisk to an IM like Jabber for presence. I believe this is already available for SER. Can anyone tell me if this is on the roadmap? I have been using both Asterisk and Jabber for quiet some time and would love to see these two working with each other. Would welcome any input on this. Shad Mortazavi Nexus Technical Manager n|m Nexus Management Inc Neutral Bay Sydney ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] How to ring internal extension?
I set it to no and that fixed my problem.. Also, I see that removing the music on hold m would have probably fixed it too. Question, isn't it a bandwidth benefit to set it to yes? What are the pros and cons? Thanks, Todd Routhier Lightwave Technologies, LLC. -- Start Your Dialup Internet Service! http://www.YourOwnISP.com Lightwave Technologies, LLC. http://www.LightWaveTech.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Senad Jordanovic Sent: Tuesday, October 19, 2004 3:26 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] How to ring internal extension? Your Own ISP .com wrote: I have it set the same for each phone within the sip.conf file if this is where you meant. FYI, I am using Grandstream 101 phones on both ends. Should it be set to yes for these phones? If you want media streams to by pass * then set it to yes. Otherwise set it to no. I would put it on no. If you opt for Yes, make sure you do not have any dial($OPTIONS). i.e. (t, r, etc/ in you Dial application. Also, make sure that codecs (especially for Grandstreams) are set in same order in sip.conf and at each phone). SJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Wonderful Success with PAP2-NA
What about all the horrible reports of humming noise on the line on many of these units? Thanks, Todd Routhier Lightwave Technologies, LLC. -- Start Your Dialup Internet Service! http://www.YourOwnISP.com Lightwave Technologies, LLC. http://www.LightWaveTech.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Boehm Sent: Tuesday, October 19, 2004 3:37 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Wonderful Success with PAP2-NA Finally got authorized to purchase some PAP2-NA's from Linksys's. Works like a charm with Asterisk. Web configuration has TONS of options and looks nice. Able to put line1 and line2 on seperate asterisk servers. Hooked a 4-line vtech phone up to 2 PAP2-NAs and basically had created a 4 line ATA for $100. -Matthew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: AW: [Asterisk-Users] Follow me using a loop
You're not going to be able to achieve what you want quite in the way you visualize it. The best advice I can give is to transfer the call to your mobile phone and take the call with you that way. This prevents the caller from being able to continue the call by simply staying on the line. On Oct 20, 2004, at 2:38 AM, Pascal C. Kocher wrote: Hello Drop the third line. exten = 31xxx,3,Goto(31xxx,1) 31xxx,2,Dial(IAX2/[EMAIL PROTECTED]CAPI/31xxx:079xxx,120,mgh) [context-Follow_me] exten = 31xxx,1,Wait(1) exten = 31xxx,2,Dial(IAX2/[EMAIL PROTECTED]CAPI/31xxx:079xxx,120,mgh) exten = i,1,hangup ; dialed an invalid number exten = t,1,hangup ; timeout By dropping the 3rd line the call does not reconnect if I (as callee) hangup the line, after the timeout the caller gets disconnected. What I am trying to achieve is what other vendors are calling mobile extension. Picking up the call on any device talking to the caller, hanging up the line and taking it (the same call) at the mobile phone to be able to walk away. All this works fine, even with moh for the caller while the call ist established again, but the problem is, that I (as callee) am unable to hangup the line. Which means, as long the caller stays on the line, the phones will keep ringing. I tried also to use any key (# or *) to hangup the call, but it seems not to dial the hangup extension. Maybe I'm just trying to achieve something weird. Best regards, Pascal. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ISDN asterisk ISDN PBX possible ?
When you say ISDN NT1, are you referring to BRI (which may use an NT1 if you need to convert a U-interface to an S/T interface) or PRI (which is also ISDN, but does not require an NT1)? Asterisk is very suitable for what you want to do, but the PRI support is far better than the BRI support. Also, you will find the price of BRI vs. PRI might not be much different on the hardware side, and PRI offers far more bandwidth. You've come to the right place. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of sjaak imap Sent: October 20, 2004 5:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] ISDN asterisk ISDN PBX possible ? Hello May be this question is a little off topic. I like to use asterisk as follow, is that possible. NT1 ISDN from telecom((isdn 1st card ) ASTERISK (2th isdn card)) excisting regular PBX system. In this way I don't have to invest in phone hardware stuff and i can join to the low cost VOIP calls. Specialy for my tele workers. Incomming calls go transparantly through the asterisk server. Outgoing calls go through asterisk to internet VOIP provider if cheaper. I'm mostly wurry about ISDN NT1 etc. Maybe someone can point me to a allready excist lowcost device on the market. Thanks Sjaak ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/ asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.775 / Virus Database: 522 - Release Date: 08/10/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.775 / Virus Database: 522 - Release Date: 08/10/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] How to ring internal extension?
Your Own ISP .com wrote: I set it to no and that fixed my problem.. Cool... :) Also, I see that removing the music on hold m would have probably fixed it too. Yap. It should of... Question, isn't it a bandwidth benefit to set it to yes? Yes... If you have no billing requirements or other services to be used from the *. What are the pros and cons? Setting it to No will almost always work... Set it to YES is a gample at the moment especially with behind NAT/ Public IP locations scenarios. SJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] How to ring internal extension?
Question, isn't it a bandwidth benefit to set it to yes? Yes... If you have no billing requirements or other services to be used from the *. Yikes, I am sure I will need to track and bill the call. This would mean I lose all of that ability? Thanks, Todd Routhier Lightwave Technologies, LLC. -- Start Your Dialup Internet Service! http://www.YourOwnISP.com Lightwave Technologies, LLC. http://www.LightWaveTech.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Senad Jordanovic Sent: Wednesday, October 20, 2004 5:48 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] How to ring internal extension? Your Own ISP .com wrote: I set it to no and that fixed my problem.. Cool... :) Also, I see that removing the music on hold m would have probably fixed it too. Yap. It should of... Question, isn't it a bandwidth benefit to set it to yes? Yes... If you have no billing requirements or other services to be used from the *. What are the pros and cons? Setting it to No will almost always work... Set it to YES is a gample at the moment especially with behind NAT/ Public IP locations scenarios. SJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] X100P problems / UK Supplier of TDM400P FXO cards
This didn't seem to appear on the list, so I'll try again. Apologies if this is a repeat post. Hi everyone, I wonder if anyone (esp. UK-based) could help us with the following (skip to the last paragraph if you don't want the detail!):- We have been trying to get an Asterisk-based VoIP server set up for our (small) office. We have two incoming analogue lines and two phones - one analogue and one multi-handset DECT. We also want to be able to send/receive calls over SIP to X-Lite. We have got a machine with Asterisk installed and SIP - SIP calls are working fine. However, we need to be able to access the two incoming analogue lines. We bought two X100P s and two GrandStream HandyTone 286s. Calls from the two POTS phones seem to be fine to SIP phones, but calls between the outside world and either the POTS phones or SIP phones are terrible. Initially we had dreadful echo, we have used 128 taps of echo cancellation and the echo is somewhat better (although it is still dreadful for the first 10-20s of the call). However, we now get quite a lot of breaking up while the person on the SIP phone / our analogue phone is talking. I believe the cause of this is the X100P - in particular, the impedance. As far as I can tell, in the UK, the lines are Zcomplex (2) = 230 nF // 1050 ohms + 320 ohms. The X100P seems to only be able to produce the US impedance = 600 ohms. As far as I can find out, the Digium TDM400P with FXO modules would not have this problem, as it has a software-adjustable impedance. However, I have not been able to find a supplier in the UK of the FXO modules for the TDM400P. So, this is where you might be able to help! * Have you used X100P cards in the UK successfully? * Do you know a supplier of TDM400P cards in the UK? Thanks Mark -- Mark Bingham Technical Director Tamsin Limited Business and Technology Centre Bessemer Drive Stevenage SG1 2DX Tel. Office: 01438 791079 Tel. Mobile: 07977177720 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] X100P problems / UK Supplier of TDM400P FXO cards
http://www.telappliant.com/ bought a lot of kit from them and would recommend them. Support line is very helpful (until you take the pee like we did :-P ). I couldn't get the X100P working even with the latest CVS build and using the apparent fix: Cidsignalling=v23 Cidstart=polarity Usecallerid=yes Try Googling site:lists.digium.com uk caller id That will show you the many many threads about this. If you do get any joy with I would be interested to know incase there is something I did wrong. HTH alex -Original Message- From: Mark Bingham [mailto:[EMAIL PROTECTED] Sent: 20 October 2004 12:03 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] X100P problems / UK Supplier of TDM400P FXO cards This didn't seem to appear on the list, so I'll try again. Apologies if this is a repeat post. Hi everyone, I wonder if anyone (esp. UK-based) could help us with the following (skip to the last paragraph if you don't want the detail!):- We have been trying to get an Asterisk-based VoIP server set up for our (small) office. We have two incoming analogue lines and two phones - one analogue and one multi-handset DECT. We also want to be able to send/receive calls over SIP to X-Lite. We have got a machine with Asterisk installed and SIP - SIP calls are working fine. However, we need to be able to access the two incoming analogue lines. We bought two X100P s and two GrandStream HandyTone 286s. Calls from the two POTS phones seem to be fine to SIP phones, but calls between the outside world and either the POTS phones or SIP phones are terrible. Initially we had dreadful echo, we have used 128 taps of echo cancellation and the echo is somewhat better (although it is still dreadful for the first 10-20s of the call). However, we now get quite a lot of breaking up while the person on the SIP phone / our analogue phone is talking. I believe the cause of this is the X100P - in particular, the impedance. As far as I can tell, in the UK, the lines are Zcomplex (2) = 230 nF // 1050 ohms + 320 ohms. The X100P seems to only be able to produce the US impedance = 600 ohms. As far as I can find out, the Digium TDM400P with FXO modules would not have this problem, as it has a software-adjustable impedance. However, I have not been able to find a supplier in the UK of the FXO modules for the TDM400P. So, this is where you might be able to help! * Have you used X100P cards in the UK successfully? * Do you know a supplier of TDM400P cards in the UK? Thanks Mark -- Mark Bingham Technical Director Tamsin Limited Business and Technology Centre Bessemer Drive Stevenage SG1 2DX Tel. Office: 01438 791079 Tel. Mobile: 07977177720 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Dear Friends of Ubiquity Software: As you may have noticed, Ubiquity Software began using the web domain ubiquity.com earlier this year in addition to the previously established ubiquity.net for our website and email communications to you. However, since that time, a dispute has emerged with respect to actual ownership of the ubiquity.com domain. As an international software company founded over decade ago, you can always reach Ubiquity Software under the website www.ubiquity.net http://www.ubiquity.net/ and via email at @ubiquity.net. However, we have also chosen to expand our domain to the more specific www.ubiquitysoftware.com http://www.ubiquitysoftware.com/ for web and @ubiquitysoftware.com for email communications. Please use either the historical ubiquity.net or begin to use the new ubiquitysoftware.com domain for all email communications to Ubiquity employees from now on. Thank you. Regards, Ubiquity Software www.ubiquitysoftware.com http://www.ubiquitysoftware.com/ [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] How to ring internal extension?
Your Own ISP .com wrote: Question, isn't it a bandwidth benefit to set it to yes? Yes... If you have no billing requirements or other services to be used from the *. Yikes, I am sure I will need to track and bill the call. This would mean I lose all of that ability? Well.. Depends what user agents you use, are there any SBC (session border controllers) used etc. Our products do deal with this issues. Please contact me of the list if you wish to hear more. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] X100P problems / UK Supplier of TDM400P FXO cards
So, this is where you might be able to help! * Have you used X100P cards in the UK successfully? YES * Do you know a supplier of TDM400P cards in the UK? As far I know TDM400P is not available from any UK supplier, because it is not approved for use in UK. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] X100P problems / UK Supplier of TDM400P FXO cards
http://www.voiptalk.org/products/Telephony+Cards/Handset+Interface+%28FX S%29+Cards?sess=982ccc6b6557552f5f60be690fde5319 TDM400P ^ As it only has FXS ports and no FXO's I guess it doesn't need to be approved as the FXS side isn't plugged directly into the PSTN ? Only a guess mind. -Original Message- From: Senad Jordanovic [mailto:[EMAIL PROTECTED] Sent: 20 October 2004 12:18 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] X100P problems / UK Supplier of TDM400P FXO cards So, this is where you might be able to help! * Have you used X100P cards in the UK successfully? YES * Do you know a supplier of TDM400P cards in the UK? As far I know TDM400P is not available from any UK supplier, because it is not approved for use in UK. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Dear Friends of Ubiquity Software: As you may have noticed, Ubiquity Software began using the web domain ubiquity.com earlier this year in addition to the previously established ubiquity.net for our website and email communications to you. However, since that time, a dispute has emerged with respect to actual ownership of the ubiquity.com domain. As an international software company founded over decade ago, you can always reach Ubiquity Software under the website www.ubiquity.net http://www.ubiquity.net/ and via email at @ubiquity.net. However, we have also chosen to expand our domain to the more specific www.ubiquitysoftware.com http://www.ubiquitysoftware.com/ for web and @ubiquitysoftware.com for email communications. Please use either the historical ubiquity.net or begin to use the new ubiquitysoftware.com domain for all email communications to Ubiquity employees from now on. Thank you. Regards, Ubiquity Software www.ubiquitysoftware.com http://www.ubiquitysoftware.com/ [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Attempt at country tones
Hi, I am attempting to put together country tones for indications.conf and zonedata.c, and hope someone can help me. 1. Are these two files the only ones that have the country tone/indications? 2. How to get my country tones included into zonedata.c, who would I send them to for inclusion? 3. Can anyone explain the meaning of dialrecall tone? Is the same tone you would get if someone put you on hold? 4. Once I get all the tones included in *, would busydetect=yes then work for me? At present, I get one minute of busy tone at the end of every voice mail because * does not recognise it, and VM only ends at the end of 1 minute, because the CO stops sending anything after 1 minute. Using an X101p card behind an ADSL micro filter, so that maybe why disconnect supervision is not recongnised?? I am a newbe, but we all need to start somewhere, and I have researched the voip-info site already. Regards Garry ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] octoBRI problem
Hi All, I have a new octoBRI card and can't seem to get it to work correctly. When I try a call from a SIP phone (Grandstream) I get nothing and when I do *CLI show channels Channel (ContextExtensionPri ) State Appl. Data 0 active channel(s) Oct 20 14:32:03 DEBUG[311315]: channel.c:464 ast_channel_walk_locked: Avoiding initial deadlock for 'Zap/11-1' Oct 20 14:32:03 DEBUG[311315]: channel.c:464 ast_channel_walk_locked: Avoiding initial deadlock for 'Zap/11-1' Oct 20 14:32:03 DEBUG[311315]: channel.c:464 ast_channel_walk_locked: Avoiding initial deadlock for 'Zap/11-1' Oct 20 14:32:03 DEBUG[311315]: channel.c:464 ast_channel_walk_locked: Avoiding initial deadlock for 'Zap/11-1' Oct 20 14:32:03 DEBUG[311315]: channel.c:464 ast_channel_walk_locked: Avoiding initial deadlock for 'Zap/11-1' Oct 20 14:32:03 DEBUG[311315]: channel.c:464 ast_channel_walk_locked: Avoiding initial deadlock for 'Zap/11-1' Oct 20 14:32:03 DEBUG[311315]: channel.c:464 ast_channel_walk_locked: Avoiding initial deadlock for 'Zap/11-1' Oct 20 14:32:03 DEBUG[311315]: channel.c:464 ast_channel_walk_locked: Avoiding initial deadlock for 'Zap/11-1' Oct 20 14:32:03 DEBUG[311315]: channel.c:464 ast_channel_walk_locked: Avoiding initial deadlock for 'Zap/11-1' Oct 20 14:32:03 DEBUG[311315]: channel.c:464 ast_channel_walk_locked: Avoiding initial deadlock for 'Zap/11-1' Oct 20 14:32:03 WARNING[311315]: channel.c:466 ast_channel_walk_locked: Avoided initial deadlock for 'Zap/11-1', 10 retries! Any hints appreciated ... George Konstantoulakis ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 2 analog phone on FXS ports?
I am a newbie on asterisk. I have a TDM400P. I want to comminicate 2 analog phones one is located in Zap/3 and another Zap/4 port. (FXS ports) When i run asterisk there will be no dial tone on the phones. Can anyone give me a small example to communicate them? (Calling each other etc. and giving them phone numbers 10 and 11) Thanks. _ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Attempt at country tones
On Wed, 20 Oct 2004, Garry Taylor wrote: At present, I get one minute of busy tone at the end of every voice mail because * does not recognise it, and VM only ends at the end of 1 minute, because the CO stops sending anything after 1 minute. Using an X101p card behind an ADSL micro filter, so that maybe why disconnect supervision is not recongnised?? Busy detection and callprogress in the chan_zap zaptel driver doesn't refer to the indications stuff. It has detection of the US tones hard-coded. Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Attempt at country tones
Garry Taylor wrote: Hi, I am attempting to put together country tones for indications.conf and zonedata.c, and hope someone can help me. 1. Are these two files the only ones that have the country tone/indications? Yes. 2. How to get my country tones included into zonedata.c, who would I send them to for inclusion? bugs.Digium.com 3. Can anyone explain the meaning of dialrecall tone? Is the same tone you would get if someone put you on hold? Sorry, no. I don't know, someone else may. 4. Once I get all the tones included in *, would busydetect=yes then work for me? Yes. Assuming they are the correct tones and cadences. Mail me off list if you would like me to analyse a voicemail recording for frequency and cadences. At present, I get one minute of busy tone at the end of every voice mail because * does not recognise it, and VM only ends at the end of 1 minute, because the CO stops sending anything after 1 minute. This will be fixed. Using an X101p card behind an ADSL micro filter, so that maybe why disconnect supervision is not recongnised?? I do the same, and have created indications for New Zealand. We now have no problems. I am a newbe, but we all need to start somewhere, and I have researched the voip-info site already. Cool. Drop me a line if you have any problems/questions, but post here if the community would benefit from the answer. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Attempt at country tones
[EMAIL PROTECTED] wrote: Busy detection and callprogress in the chan_zap zaptel driver doesn't refer to the indications stuff. It has detection of the US tones hard-coded. busydetect=yes and busycount=10 (or whatever) definitely refer to the indications. These are not hard coded for the US. Hence the fact they work here in New Zealand with totally different tones. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Samsung DCS70 PABX
Hi there, I'm currently working with a customer who has 25 Phones as part of their exisiting Telephone System which runs using a Samsung DCS70 PABX. They currently have a Single PRI coming in with a block of 100 Phone Numbers. I'm trying to connect them to our SIP Proxy so that we can connect their multiple sites together and provide a local telephone system using VoIP. After a search on google, I don't believe this PABX supports VoIP directly on it. Is there a way that anybody can recommend by which they don't have to phase out their PABX but can also make use of our SIP Proxy for Outgoing Calls? Cheers, Sahil ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] CAPI and Asterisk (with AVM ISDN Card)
Thank you for your help ;-) I have downloadet from the url you gave me the following file. isdn4k-utils-CVS-2004-10-07.tar.bz2 When I cd into that folder.. and type make conf I get a wirred screen where I can choose from the following: Code maturity level options --- General configuration --- Runtime configuration tools --- Card configuration tools --- Tools for monitoring activity --- Applications --- Documentation --- --- Load an Alternate Configuration File Save Configuration to an Alternate File Any idears ? I spend so hours on this now.. Do you know anyone ( or do you) offer a service where I can pay someone to login and help me install it ? ( capi + my card ) Thank you Grsse / Best Regards Mateo Meier --- Don't marry for money; you can borrow it cheaper ;-) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Maurizio Marini Sent: Mittwoch, 20. Oktober 2004 09:04 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] CAPI and Asterisk (with AVM ISDN Card) On Wednesday 20 October 2004 00:30, Mateo Meier wrote: Does anybody knows what version of capi is needed ? try the most recent here: ftp://ftp.isdn4linux.de/pub/isdn4linux/CVS-Snapshots it did work fine for me (FC2 and debian sid) I tried to install a capi rpm.. but after the capi rpm installation, there seems to be no /etc/capi.conf cat capi.conf # card fileproto io irq mem cardnr options b1isa b1.t4 DSS10x150 7 - - P2P b1pci b1.t4 DSS1- - - - c4 /usr/sbin/c4.binDSS1- - - - c4 - DSS1- - - - c4 - DSS1- - - - P2MP c4 - DSS1- - - - P2MP c2 c2.bin DSS1- - - - c2 - DSS1- - - - t1isa t1.t4 DSS10x340 9 - 0 t1pci t1.t4 DSS1- - - - fcpci - - - - - - fcclassic - - 0x150 10 - - What kind of capi version do I need ? capi4k-utils ? or just any capi rpm ? download a tarball and install it... Maurizio ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] cheap gig switch? smc, netgear, or 3com?
I have one of these, works great but failed about 6 months into it's life, was replaced on the spot (in Australia (I'm originally from there) but you had to drive it to them with the original receipt for the handover). Does anyone know if this is a worldwide warranty? Has anyone in NY tried to claim? Where was it etc? Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jay Wilton Sent: Wednesday, October 20, 2004 4:08 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] cheap gig switch? smc, netgear, or 3com? Hello, The Smc 8508T goes for about $95, jumbo frame support, lifetime warranty but no QOS. The Netgear GS608 is $ 100, no jumbo frames, 1 year warranty, QOS, gig latency 10U max. The 3com switch reviews that I read were not happy. Does anyone hate or love their home switch? I doubt the jumbo frame support would help voip traffic, but it seems like it wouldn't hurt. I was planning on doing the QOS on linux. Gig support is wanted for file transfers and the future. Thanks to all you nice asterisk people and a few of the mean ones. Jay __ Do you Yahoo!? Yahoo! Mail Address AutoComplete - You start. We finish. http://promotions.yahoo.com/new_mail ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cannot call OH323 swissvoice Phone
Hi all,
I have completed asterisk-oh323 version 0.5.10 and I've registered it in
Gatekeeper (Cisco 3640 wich is H323 Proxy with Gatekeeper features), I've
also registered a Swissvoice in Gatekeeper . Now, when I make calls from
Cisco it works fine ,
but when I try to call from X-Lite it shows me the following errors :
Executing Dial(SIP/310-2dc9, OH323/[EMAIL PROTECTED]) in new stack
Oct 20 14:50:12 ERROR[360471]: chan_oh323.c:2631 setup_h323_connection:
Request to open an existing channel 0 with the same direction 1.
-- Called [EMAIL PROTECTED]
Oct 20 14:50:12 WARNING[327701]: chan_oh323.c:1400 oh323_read: OH323/L20192:
Invalid format of RTP addresses.
-- Hungup 'OH323/L20192'
== No one is available to answer at this time
My oh323.conf is :
; Configuration file of OpenH323 channel driver
[general]
listenAddress=0.0.0.0
;
listenPort=1720
;
;
connectPort=1720
;
tcpStart=1
tcpEnd=2
;
udpStart=1
udpEnd=2
;
;
fastStart=no
;
h245Tunnelling=no
;
;
h245inSetup=no
;
inBandDTMF=no
;
silenceSuppression=yes
;
jitterMin=20
jitterMax=1000
;
ipTos=lowdelay
;
;
outboundMax=10
inboundMax=10
simultaneousMax=10
;
;
bandwidthLimit=1024
;
wrapLibTraceLevel=1
libTraceLevel=0
libTraceFile=stdout
;
gatekeeper=10.1.0.51
;
;
gatekeeperTTL=600
;
userInputMode=TONE
;-
; Configure H.323 aliases, prefixes and
; related ASTERISK's contexts
;-
[register]
;
context=h323
alias=astra
;-
; Specify and configure CODEC related
; options
;-
[codecs]
;
;
codec=G711A
frames=20
;
[astra]
type=h323
prefix=400
context=h323
My extension.conf is:
[general]
static=yes
writeprotect=no
[h323]
exten = 400,1,Dial(OH323/[EMAIL PROTECTED])
include = sip
include = mgcp
[mgcp]
exten = 411,1,Dial(MGCP/aaln/[EMAIL PROTECTED])
include = h323
include = sip
[sip]
include = mgcp
include = h323
exten = _[3]XX,1,NoOp(^D3call for ^D3${EXTEN})
exten = _[3]XX,2,Dial(SIP/${EXTEN},60,tr)
exten = _[3]XX,3,Congestion()
I can see that asterisk is registered in gatekeeper
*CLI oh323 show conf
Configuration of OpenH323 channel driver
Version: 0.5.10
Listening on address: 0.0.0.0:1720
Gatekeeper used: [EMAIL PROTECTED]
FastStart/H245Tunnelling/H245inSetup: OFF/OFF/OFF
Supported format(s): ALAW0
Jitter buffer limits (min/max): 20-1000 ms
TCP port range: 1 - 2
UDP (RAS) port range: 1 - 2
UDP (RTP) port range: 1 - 2
IP Type-of-Service value: 16
User input mode: 2
Max number of inbound H.323 calls: 10
Max number of outbound H.323 calls: 10
Max number of simultaneous H.323 calls: 10
Anyone to help me ???
Regards,
Astrit Morina
System Operator
Tel: 038 20304050
Fax: 038 20304020
E-mail: [EMAIL PROTECTED]
www.ipko.net
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[Asterisk-Users] New Channel Driver: chan_bluetooth
Hi, after a couple of days work banging my head against the wall (bloody standards my arse), i've got chan_bluetooth to a point where it's starting to function - certianly more than just proof of concept now. The code is far, far from stable, and only been tested with one dongly (Cambridge Silicon), and two devices (Nokia 6310i and a HBH-200 headset) - and even those devices are not FULLY functional yet. I can make and recieve calls through the devices, at times though ;) So why am I announcing it when it doesn't even work properly? I'd like to get people testing it with their devices and bluetooth adapters, see if it will pair, see debug output - see what phones/headsets do and don't work, see what documentation i need to add for setting it up, and of course get some general feedback about peoples wishes/views on the module. Code is at http://www.crazygreek.co.uk/chan_bluetooth Follow instructions on that page, or ask on list [1] when you encounter problems! Please, let me have some feedback so i can get this driver working properly with all HeadsetProfile devices! ~ Theo [1] - http://tribble.crazygreek.co.uk/cgi-bin/mailman/listinfo/chan_bluetooth -- Theo P. Zourzouvillys [EMAIL PROTECTED] http://www.crazygreek.co.uk/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] X100P problems / UK Supplier of TDM400P FXO cards
Alex Barnes wrote: http://www.voiptalk.org/products/Telephony+Cards/Handset+Interface+%28FX S%29+Cards?sess=982ccc6b6557552f5f60be690fde5319 TDM400P ^ As it only has FXS ports and no FXO's I guess it doesn't need to be approved as the FXS side isn't plugged directly into the PSTN ? No it does not, but the question was about FXO ports not FXS. When I spoke to Telappliant about 2 months ago, they said FXO modules are not CE approved. That still stands today. However, you can buy FXO module from them for testing purpose only in which case there is no support provided. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Samsung DCS70 PABX
Should be easy enough. It is especially simple if they have enough bandwidth at each site. You can add additional PRI cards in the Samsungs and connect those to Asterisk servers. Then just used LCR in the Samsung to send dialed calls to appropriate trunks connected to asterisk. Then just set the Dial plan in asterisk to send the call to the appropriate place-the other asterisk servers. The other asterisk will use the DID capability to send the call to the appropriate phones. I have this set up using Toshiba PBX's in my lab and works great. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Wednesday, October 20, 2004 7:15 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Samsung DCS70 PABX Hi there, I'm currently working with a customer who has 25 Phones as part of their exisiting Telephone System which runs using a Samsung DCS70 PABX. They currently have a Single PRI coming in with a block of 100 Phone Numbers. I'm trying to connect them to our SIP Proxy so that we can connect their multiple sites together and provide a local telephone system using VoIP. After a search on google, I don't believe this PABX supports VoIP directly on it. Is there a way that anybody can recommend by which they don't have to phase out their PABX but can also make use of our SIP Proxy for Outgoing Calls? Cheers, Sahil ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cannot call OH323 swissvoice Phone
Sorry , there was a mistake , the correction is (I wrote logs when I had
gatekeeper 10.1.0.50 , but now is 10.1.0.51):
Executing Dial(SIP/310-2dc9, OH323/[EMAIL PROTECTED]) in new stack Oct 20
14:50:12 ERROR[360471]: chan_oh323.c:2631 setup_h323_connection:
Request to open an existing channel 0 with the same direction 1.
-- Called [EMAIL PROTECTED]
Oct 20 14:50:12 WARNING[327701]: chan_oh323.c:1400 oh323_read: OH323/L20192:
Invalid format of RTP addresses.
-- Hungup 'OH323/L20192'
== No one is available to answer at this time
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Astrit
Sent: Wednesday, October 20, 2004 2:39 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] Cannot call OH323 swissvoice Phone
Hi all,
I have completed asterisk-oh323 version 0.5.10 and I've registered it in
Gatekeeper (Cisco 3640 wich is H323 Proxy with Gatekeeper features), I've
also registered a Swissvoice in Gatekeeper . Now, when I make calls from
Cisco it works fine ,
but when I try to call from X-Lite it shows me the following errors :
Executing Dial(SIP/310-2dc9, OH323/[EMAIL PROTECTED]) in new stack Oct 20
14:50:12 ERROR[360471]: chan_oh323.c:2631 setup_h323_connection:
Request to open an existing channel 0 with the same direction 1.
-- Called [EMAIL PROTECTED]
Oct 20 14:50:12 WARNING[327701]: chan_oh323.c:1400 oh323_read: OH323/L20192:
Invalid format of RTP addresses.
-- Hungup 'OH323/L20192'
== No one is available to answer at this time
My oh323.conf is :
; Configuration file of OpenH323 channel driver
[general]
listenAddress=0.0.0.0
;
listenPort=1720
;
;
connectPort=1720
;
tcpStart=1
tcpEnd=2
;
udpStart=1
udpEnd=2
;
;
fastStart=no
;
h245Tunnelling=no
;
;
h245inSetup=no
;
inBandDTMF=no
;
silenceSuppression=yes
;
jitterMin=20
jitterMax=1000
;
ipTos=lowdelay
;
;
outboundMax=10
inboundMax=10
simultaneousMax=10
;
;
bandwidthLimit=1024
;
wrapLibTraceLevel=1
libTraceLevel=0
libTraceFile=stdout
;
gatekeeper=10.1.0.51
;
;
gatekeeperTTL=600
;
userInputMode=TONE
;-
; Configure H.323 aliases, prefixes and
; related ASTERISK's contexts
;-
[register]
;
context=h323
alias=astra
;-
; Specify and configure CODEC related
; options
;-
[codecs]
;
;
codec=G711A
frames=20
;
[astra]
type=h323
prefix=400
context=h323
My extension.conf is:
[general]
static=yes
writeprotect=no
[h323]
exten = 400,1,Dial(OH323/[EMAIL PROTECTED]) include = sip include = mgcp
[mgcp]
exten = 411,1,Dial(MGCP/aaln/[EMAIL PROTECTED]) include = h323 include = sip
[sip]
include = mgcp
include = h323
exten = _[3]XX,1,NoOp(^D3call for ^D3${EXTEN}) exten =
_[3]XX,2,Dial(SIP/${EXTEN},60,tr) exten = _[3]XX,3,Congestion()
I can see that asterisk is registered in gatekeeper
*CLI oh323 show conf
Configuration of OpenH323 channel driver
Version: 0.5.10
Listening on address: 0.0.0.0:1720
Gatekeeper used: [EMAIL PROTECTED]
FastStart/H245Tunnelling/H245inSetup: OFF/OFF/OFF Supported format(s):
ALAW0 Jitter buffer limits (min/max): 20-1000 ms TCP port range: 1 -
2 UDP (RAS) port range: 1 - 2 UDP (RTP) port range: 1 -
2 IP Type-of-Service value: 16 User input mode: 2 Max number of inbound
H.323 calls: 10 Max number of outbound H.323 calls: 10 Max number of
simultaneous H.323 calls: 10
Anyone to help me ???
Regards,
Astrit Morina
System Operator
Tel: 038 20304050
Fax: 038 20304020
E-mail: [EMAIL PROTECTED]
www.ipko.net
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RE: [Asterisk-Users] Samsung DCS70 PABX
You've come to the right place :-) From what you describe, you'd want to do something along these lines: [PSTN][Asterisk][Existing PBX] | | [SIP Proxy] You'd need two T1 cards in the Asterisk to make this work, one to connect to the PSTN, the other to connect to the PBX. Alternatively, you might be able to do this [PSTN]---[existing PBX]---[Asterisk]---[SIP proxy] In this case you'd only need one T1 card in the Asterisk, but you'd have to add another PRI card to the PBX (assuming it supports more than one T1 card). -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: October 20, 2004 8:15 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Samsung DCS70 PABX Hi there, I'm currently working with a customer who has 25 Phones as part of their exisiting Telephone System which runs using a Samsung DCS70 PABX. They currently have a Single PRI coming in with a block of 100 Phone Numbers. I'm trying to connect them to our SIP Proxy so that we can connect their multiple sites together and provide a local telephone system using VoIP. After a search on google, I don't believe this PABX supports VoIP directly on it. Is there a way that anybody can recommend by which they don't have to phase out their PABX but can also make use of our SIP Proxy for Outgoing Calls? Cheers, Sahil ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/ asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.775 / Virus Database: 522 - Release Date: 08/10/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.775 / Virus Database: 522 - Release Date: 08/10/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Samsung DCS70 PABX
If it has a spare PRI port then build up an * server with an E100 card and connect using an ISDN crossover cable. If the Samsung can support analog trunks you could stick in a TDM400 with a couple of FXO ports. Craig - Original Message - From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, October 20, 2004 8:14 PM Subject: [Asterisk-Users] Samsung DCS70 PABX Hi there, I'm currently working with a customer who has 25 Phones as part of their exisiting Telephone System which runs using a Samsung DCS70 PABX. They currently have a Single PRI coming in with a block of 100 Phone Numbers. I'm trying to connect them to our SIP Proxy so that we can connect their multiple sites together and provide a local telephone system using VoIP. After a search on google, I don't believe this PABX supports VoIP directly on it. Is there a way that anybody can recommend by which they don't have to phase out their PABX but can also make use of our SIP Proxy for Outgoing Calls? Cheers, Sahil ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] X100P problems / UK Supplier of TDM400P FXO cards
Hi, Voiptalk and telepliant are one and the same company. I spoke to their support a while back and they say they are waiting approval for the FXO modules. I bought my TDM01B direct from Digium, took about 2 weeks to arrive in the UK and has fixed all of my impedence problems as well as supporting polarity callerid. Regards Ian From: Alex BarnesSent: Wed 20/10/2004 12:21To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [Asterisk-Users] X100P problems / UK Supplier of TDM400P FXO cards http://www.voiptalk.org/products/Telephony+Cards/Handset+Interface+%28FX S%29+Cards?sess=982ccc6b6557552f5f60be690fde5319 TDM400P ^ As it only has FXS ports and no FXO's I guess it doesn't need to be approved as the FXS side isn't plugged directly into the PSTN ? Only a guess mind. -Original Message- From: Senad Jordanovic [mailto:[EMAIL PROTECTED] Sent: 20 October 2004 12:18 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] X100P problems / UK Supplier of TDM400P FXO cards So, this is where you might be able to help! * Have you used X100P cards in the UK successfully? YES * Do you know a supplier of TDM400P cards in the UK? As far I know TDM400P is not available from any UK supplier, because it is not approved for use in UK. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Dear Friends of Ubiquity Software: As you may have noticed, Ubiquity Software began using the web domain ubiquity.com earlier this year in addition to the previously established ubiquity.net for our website and email communications to you. However, since that time, a dispute has emerged with respect to actual ownership of the ubiquity.com domain. As an international software company founded over decade ago, you can always reach Ubiquity Software under the website www.ubiquity.net http://www.ubiquity.net/ and via email at @ubiquity.net. However, we have also chosen to expand our domain to the more specific www.ubiquitysoftware.com http://www.ubiquitysoftware.com/ for web and @ubiquitysoftware.com for email communications. Please use either the historical ubiquity.net or begin to use the new ubiquitysoftware.com domain for all email communications to Ubiquity employees from now on. Thank you. Regards, Ubiquity Software www.ubiquitysoftware.com http://www.ubiquitysoftware.com/ [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Mail was checked for spam by the Freeware Edition of No Spam Today! The Freeware Edition is free for personal and non-commercial use. You can remove this notice by purchasing a full license! To order or to find out more please visit: http://www.no-spam-today.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Wonderful Success with PAP2-NA
Call linksys and tell them you want to become authorized. They will ask which of their 3 distros you have an account with. Stay on them. Call each day. They won't do anything unless you stay on them. Matthew - Original Message - From: Bartosz Jozwiak [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Tuesday, October 19, 2004 6:11 PM Subject: Re: [Asterisk-Users] Wonderful Success with PAP2-NA How to get NA version from Linksys. We are ISP and VoIP provider and would like to sell these box'es to our local customers. I have mailed sales at linksys but no reply. Could somebody tell me how to get approved to buy NA version of these boxes. Thank you in advance! Bartek - Original Message - From: Kevin P. Fleming [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Tuesday, October 19, 2004 7:33 PM Subject: Re: [Asterisk-Users] Wonderful Success with PAP2-NA Gonzalo Servat wrote: What is this huge OH MY GOD difference between the two? (apart from the -NA). I've googled and can't seem to find any site that lists the difference(s). The non-NA version is for Vonage service only, and comes out of the box locked. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] grandstream handytone 286 problem
hey, i got asterisk running with softphones but now I received a set of grandstream handytone 286's now if I run the setup and configure everything like supposed to be it doesn't work, i hear a ringing tone , after 30secs it hangs up and that's it in sip.conf i have: [4445] secret=4445 type=friend username=christophe allow=all host=10.0.0.55 nat=yes [] secret= type=friend username=nicole allow=all host=10.0.0.56 nat=yes and for configuration of my grandstream handytone 286 i got: sip server: 10.0.0.21 sip user id: christophe authenticate id: 4445 authenticate password: 4445 and as vocoder i got: G729 G729 G729 G729 PCMU PCMA PCMU sip registration: yes unregister at reboot: yes anyone know what i could be doing wrong ? -- Christophe De Coninck | Zarek K ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] X100P problems / UK Supplier of TDM400P FXO cards
Senad Jordanovic [EMAIL PROTECTED] wrote: Alex Barnes wrote: TDM400P As it only has FXS ports and no FXO's I guess it doesn't need to be approved as the FXS side isn't plugged directly into the PSTN? No it does not, but the question was about FXO ports not FXS. When I spoke to Telappliant about 2 months ago, they said FXO modules are not CE approved. That still stands today. However, you can buy FXO module from them for testing purpose only in which case there is no support provided. That testing would have to be performed on a private telephone network, of course. It is still strictly illegal to connect the FXO modules to a public telephone network in England, and most likely anywhere in Europe as well, as far as I know. I'm sure someone will rush to correct me if I'm wrong. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Attempt at country tones
Garry Taylor wrote: 2. How to get my country tones included into zonedata.c, who would I send them to for inclusion? Open a bug report and add a patch to the bug tracker, http://bugs.digium.com /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP phones
Title: SIP phones I am looking for a loud ringing SIP phone. I am presently using the Polycom and just cannot loud enough to hear it over the din in a collocation room. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP phones
Title: SIP phones Why dont you use an ATA device with a loud regular phone and/or hook up one of those really loud ringing devices you can get at a phone shop? J Just a suggestion. S. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of Michael Di Martino Sent: Wednesday, October 20, 2004 6:40 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] SIP phones I am looking for a loud ringing SIP phone. I am presently using the Polycom and just cannot loud enough to hear it over the din in a collocation room. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] still riniging problem
Good day all I have a problem with the new asterisk version.My setup,extensions.conf is like this: If someone call in from the outside to the PSTN,asterisk wait 8s and then forwards the call to the sip user,the operator and she then transfer calls My problem is,for the first 8s the ringing sound is normal but as soon as the call goes to the sipuser(operator) the ringing gets very fast and and some people thinks its a busy signal Previous versions worked Please help Altus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Vmail.cgi Bahhh!!
OK, been at this for a few hours now. I am on Fedora 2 trying like heck to get the web based Vmail thing working. I have it to the point where I can login to it successfully but no messages ever show up there even though I know they exist. I am getting the voicemail messages OK in my email. I CHMOD'd the vmail dir to 777 just for testing purposes, I installed the required Perl Module. Not sure what's left to try, any ideas would be most welcome. Thanks, Todd Routhier Lightwave Technologies, LLC. -- Start Your Dialup Internet Service! http://www.YourOwnISP.com Lightwave Technologies, LLC. http://www.LightWaveTech.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] cannot call Grandstream
I am having trouble with a Grandstream Budgetone 101. It's at firmware 1.0.5.10 and I'm running * 1.0.0. I have the phone getting a DHCP address and * expects it to register. When I reboot the phone it does register just fine. However, after a while * cannot contact the phone. I will call the phone and * will tell me: -- Called grandstream1 Oct 20 09:41:16 WARNING[98310]: chan_sip.c:681 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Critical Request) Looking in teh archives, it seems that that indicates that the registration is expired. I've got the phone set to 60m register intervals (and * acks that when the phone registers) but after the hour it doesn't re-register. I've also tried 15m and 2m register timeouts. I have Sip Registration and Unregister on Reboot both set to Yes on the phone. Register Expiration is 60. The phone is at 192.168.42.234 and * is as 192.168.1.3. Both internal but no NAT between them. And the initial registration works fine. I've searched through the mail list archives and tried all the suggestions I could find there, but the phone behaves the same: registration appears to be lost. Incidentally, I set the phone to a static IP (192.168.42.99) and also set * from host=dynamic to host=192.168.42.99 but * couldn't call the phone at all after that. (I did graceful restarts on * between the change). Can anyone see what I might be missing? I don't have the SIP UserID or Authenticate ID set to the phone's extension, but the SIP User ID is the same as the Authenticate ID which is the same as the context in *'s sip.conf. It doesn't seem that would have an effect, but I thought I'd mention it. Thanks! -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fax over IP doesn't works
Hi Miroslav, It sounds like you don't really understand what T.38 is. You need some form of modem to get from a normal FAX machine to a T.38 channel. spandsp can do that. A normal FAX modem can do that. You need a modem somewhere, though. That is why developing the FAX modems was the first step towards providing T.38. Regrds, Steve Miroslav Nachev wrote: Dear Steve, SU So how does the FAX get from the fax machine to the T.38 channel SU with spandsp? In our case we will try to strip spandsp and will use directly OpenH323. We do tests for compatibility with one of the biggest national telecom and if they are OK, they will offer Asterisk based IP PBX to their clients instead Cisco. That's why we need of T.38 and G.711 fax capabilities. Also we have the problems with the following tests: 1. When Dialing of unallocated number the resposne must be Invalid Number, but the result is one of the following: Hangup, Congestion or Busy. 2. CLIP/CLIR User provided verified and passed - We can't find where we can set this bits for this services. 3. Fax T38 / g711 4. Codec negotiation: when 2 codecs are possible (G.711 and G.729), the two parties can't negotiate which codec to use. Best Regards, Miroslav Nachev Miroslav Nachev wrote: SU and exactly how does that get the FAX into the T.38 channel? :-\ Using G.711 or implementing T.38 in Asterisk or adjusting Asterisk to OpenH323 T.38. From our expirience Asterisk detect that the line is with Fax data. The problem is what next. So how does the FAX get from the fax machine to the T.38 channel with spandsp? Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] [OT] GSM patents
Hi guys, Where can I find information about GSM codec patents, fees or other legal information? I'm interested in develop a VoIP software and I'd like to use GSM codec. Thanks in advance, Fede ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Samsung DCS70 PABX
Hi, I didn't actually realise I could quite simply just use an Asterisk Server with FXS Modules. The final issue in planning this setup is that we want to send all local and mobile calls out back via the PRI and all other toll calls out via the FXS Lines. I've never worked with Hardware Based PBX's, do they normally have dial plans similar to the way Asterisk Operates? Thanks all for replying to my post. Cheers, Sahil Quoting Jim Van Meggelen [EMAIL PROTECTED]: You've come to the right place :-) From what you describe, you'd want to do something along these lines: [PSTN][Asterisk][Existing PBX] | | [SIP Proxy] You'd need two T1 cards in the Asterisk to make this work, one to connect to the PSTN, the other to connect to the PBX. Alternatively, you might be able to do this [PSTN]---[existing PBX]---[Asterisk]---[SIP proxy] In this case you'd only need one T1 card in the Asterisk, but you'd have to add another PRI card to the PBX (assuming it supports more than one T1 card). -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: October 20, 2004 8:15 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Samsung DCS70 PABX Hi there, I'm currently working with a customer who has 25 Phones as part of their exisiting Telephone System which runs using a Samsung DCS70 PABX. They currently have a Single PRI coming in with a block of 100 Phone Numbers. I'm trying to connect them to our SIP Proxy so that we can connect their multiple sites together and provide a local telephone system using VoIP. After a search on google, I don't believe this PABX supports VoIP directly on it. Is there a way that anybody can recommend by which they don't have to phase out their PABX but can also make use of our SIP Proxy for Outgoing Calls? Cheers, Sahil ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/ asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.775 / Virus Database: 522 - Release Date: 08/10/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.775 / Virus Database: 522 - Release Date: 08/10/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone else seeing this?
Hi Brian, I haven't reported this yet, as I don't have an overall picture of what is happening, but A couple of weeks ago I had several machine lockups on the same day while testing MFC/R2 with a tor2. It hasn't happened any more here. I have no idea why it suddenly started or stopped. However, now people are starting to deploy R2, I have reports of occasional lockups with tor2 cards. I have no idea if these lockups have the same cause as mine. Regards, Steve Brian West wrote: Anything after these versions: zaptel.c version 1.95 (known working) chan_zap.c version 1.357 (known working) with a tor2 card... causes kernel panic... Can anyone else confirm this? I honestly think it's a combo issue with the new zap reload and that zaptel change. But I have spent hours trying to narrow it down to those two files and those changes. Has anyone else seen strange issues when using PRI? If we have zaptel.c 1.95 and latest chan_zap.c you can place and take calls but if you do something like show channels at the CLI you'll deadlock the box. I have no thread apply all bt since the glibc on this box didn't have debug compiled in on it. (will retry this tomorrow) If you have the latest zaptel.c and the latest chan_zap.c placing any call out/in the zap interface will cause a kernel panic and kill the box. Use the above listed known working files and you have no problems. I would open a bug report but I would like to find more information before doing so. Thanks, Brian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] i extension
Benjk, Good question, I develop asterisk code and did not know the i extension existed for some time. The wiki is very poor on it. If you put the I extension in a context with the dial plan then it works best. This way the problem is handled locally within the dial plan instead of fall through to the global context. [context123] exten = _222XXX,1,Dial(...) ... exten = i,1,Fix The problem for invalid extension in dial plan. ... Exten = t,1,Fix The problem for timeout in dial plan ... [context124] ... You could create a [invalid-Extension-Fix-Context] and do an include in the other contexts. Race Vanderdecken AsTerisk 8-e Vanderdecken DOT coM -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Benjamin on Asterisk Mailing Lists Sent: 19 October 2004 17:23 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] i extension This is a question I am almost too embarassed to ask but here we go ... Is it not possible to use the i extension to trap attempts of users misdialling numbers otherwise not in the dialplan/context? I have seen this in so many examples and I always thought Oh, this will come in handy one day but never actually had to use it. Now, as I have a customer who is complaining every day about the Asterisk server not working because they seem to have fingers to thick for their phones' keypads or suffer from some rare form of number dislexia, I would really like to trap all those cases where they dial some invalid number that's not matching anything in the dialplan/context and play an invalid number/extension recording back to the user. Unfortunately, it doesn't seem to work and I couldn't find anything other than the Wiki page that also says it doesn't work but fails to explain why. So, what's the story on this? thanks rgds benjk -- Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Tokyo, Japan. NB: Spam filters in place. Messages unrelated to the * mailing lists may get trashed. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] [OT] GSM patents
Federico Edelman wrote: Hi guys, Where can I find information about GSM codec patents, fees or other legal information? I'm interested in develop a VoIP software and I'd like to use GSM codec. Thanks in advance, Fede Where would be a good place to look for patents on ETSI specs? www.etsi.org maybe? :-) They do, in fact, list all the people who claim IP rights over each spec. They don't spell out all the details, though. I doubt there are simple licencing programs for using these things in anything but a GSM or UMTS system. That is the only place most of the codecs have been deployed. The original 06.10 codec, which * supports, seems to be free of patent problems. Its just too old to have any. :-) Philips used to claim one, but I think its safe now. Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Alternatives to the T100Ps?
- Original Message - From: Michael Loftis [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Tuesday, October 19, 2004 6:14 PM Subject: RE: [Asterisk-Users] Alternatives to the T100Ps? | | | --On Sunday, October 17, 2004 10:31 -0400 Brian Kurkowski | [EMAIL PROTECTED] wrote: | | Michael, | | I usually read and don't do much posting, but I had to on this. | | Sorry things getting badly buried lately recent reply brought this | thread back to my attention and I realised I'd missed this post. | | I am really suprised to see your commnets, and wondered what is the basis | ? We have had a dual Xenon with a quad port T1 card in production for 16 | months processing as many as 20,000 messaging calls a day. The box has | never crashed, the board has never crashed, we haven't even restarted | asterisk much less upgraded the code. I have never take a Bit Error on my | DMS-500 from a Digium card. This is only one of several production | boxes but the story is the same on all of them. | | How in the heck does this equate to: hardware, drivers, or both is | pretty sketchy ? | | The fact that they are REALLY picky on what they work in, and they either | work really well (as in your case) or (as in my example) cause the system | to go totally flake when it's otherwise been known to run excellently in | all situations. Whether it's hardware being picky, drivers being somewhat | bad behavior or something else I'm not sure. The 1kHz clock that they keep | should be easily followed by any modern hardware -- I've built applications | based around faster interrupt rates on less hardware (Intel and AMD based). | | I would suggest just the opposite. Mark and the boys have done a great job | on all fronts. How many Cisco AS-5300's have that record ? I have 9 of | them brand new and not a single one is my answer. | | There's no doubt that Digium brought this card into mass production, | cleaned it up, improved upon it, and have done so steadily since it's | creation. I also have no doubt whatsoever that they will continue to do | so, and very aggressively. We'll probably also start to see more products | coming from them, I have no idea what but they have a lot of smart folks | over there. | | | If you haven't looked at Digium lately, look again. | | I've got two of their cards right now :) That's what sparked the whole | thread. | | I don't mind paying more, neither do most businesses, for hardware that's | more solid, or handles a given task better. I suppose with the Digium | boards I could dive into the VHDL and reprogram the FPGA if I find any | problems. I just don't know the current state of all of those bits. | | The PBX we've built seems ot be very stable in it's new motherboard, but | it's still very...curious that it behaved so badly in a known good | motherboard with more than enough horsepower -- 1.4Ghz clock -- AMD Athlon | 1800+, w/ 1.5Gig of RAM, all clean, tested pretty regularly with memtest86 | and other diagnostics as I use it for a bench machine. | | Though I think it more likely had more to do with some unhealthy | interaction on the motherboard and card rather than one or the other, which | seems to be reported occasionally by T100P buyers, and the TDM400P also | seems to have some similar issues. | | Now the fact that there are so many configurations under which the T100P | and TDM400P work VERY well means that the fundamentals are absolutely | right, there's just some sort of edge case. I just happen to be of the | opinion that the real world is an edge case so if you can't handle a fairly | common COTS setup like the system I described above, then there's something | that needs some pretty good improvement somewhere. | | The whole thing is just my opinions and thoughts, and tempered by the | (relatively bad) experience I had getting these cards going because they | just would not play with any motherboard I threw at them until we went for | rather top of the line motherboard. | Michael, are you successfully routing data over the t100p card as well as voice? Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Vmail.cgi Bahhh!!
On Oct 20, 2004, at 8:49 AM, Your Own ISP .com wrote: OK, been at this for a few hours now. I am on Fedora 2 trying like heck to get the web based Vmail thing working. I have it to the point where I can login to it successfully but no messages ever show up there even though I know they exist. I am getting the voicemail messages OK in my email. I CHMOD'd the vmail dir to 777 just for testing purposes, I installed the required Perl Module. Not sure what's left to try, any ideas would be most welcome. Thanks, Todd Routhier Lightwave Technologies, LLC. Todd, asterisk records the files to the filesystem with root permissions, which a properly configured apache installation doesn't have access too. I worked around this by setting up a cronjob to chmod 777 all the voicemail files once a minute, which probably isn't the most elegant solution to this problem. my crontab entry: # cheap way to fix our permissions for voicemail * * * * * /etc/vm_chmod.bat /dev/null /etc/vm_chmod.bat: #!/bin/sh chmod -R 777 /var/spool/asterisk/vm If someone else doesn't give a better solution, you can try this. Niles ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re[2]: [Asterisk-Users] Fax over IP doesn't works
Dear Steve, I can't understand from your mail can I use SpanDSP or not? Today we try this fax-modem: http://www.openh323.org/t38.html The problem now is that we can't start it with HylaFAX. Best Regards, Miroslav Nachev SU Hi Miroslav, SU It sounds like you don't really understand what T.38 is. You need some SU form of modem to get from a normal FAX machine to a T.38 channel. SU spandsp can do that. A normal FAX modem can do that. You need a modem SU somewhere, though. That is why developing the FAX modems was the first SU step towards providing T.38. SU Regrds, SU Steve SU Miroslav Nachev wrote: Dear Steve, SU So how does the FAX get from the fax machine to the T.38 channel SU with spandsp? In our case we will try to strip spandsp and will use directly OpenH323. We do tests for compatibility with one of the biggest national telecom and if they are OK, they will offer Asterisk based IP PBX to their clients instead Cisco. That's why we need of T.38 and G.711 fax capabilities. Also we have the problems with the following tests: 1. When Dialing of unallocated number the resposne must be Invalid Number, but the result is one of the following: Hangup, Congestion or Busy. 2. CLIP/CLIR User provided verified and passed - We can't find where we can set this bits for this services. 3. Fax T38 / g711 4. Codec negotiation: when 2 codecs are possible (G.711 and G.729), the two parties can't negotiate which codec to use. Best Regards, Miroslav Nachev Miroslav Nachev wrote: SU and exactly how does that get the FAX into the T.38 channel? :-\ Using G.711 or implementing T.38 in Asterisk or adjusting Asterisk to OpenH323 T.38. From our expirience Asterisk detect that the line is with Fax data. The problem is what next. So how does the FAX get from the fax machine to the T.38 channel with spandsp? Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] grandstream handytone 286 problem
update: now I get this after i repowered the grandstream handytone 286 : *CLI Oct 20 16:34:33 NOTICE[262160]: chan_sip.c:7532 handle_request: Registration from 'sip:[EMAIL PROTECTED];user=phone' failed for '10.0.0.55 On Wed, 2004-10-20 at 15:30, christophe de coninck wrote: hey, i got asterisk running with softphones but now I received a set of grandstream handytone 286's now if I run the setup and configure everything like supposed to be it doesn't work, i hear a ringing tone , after 30secs it hangs up and that's it in sip.conf i have: [4445] secret=4445 type=friend username=christophe allow=all host=10.0.0.55 nat=yes [] secret= type=friend username=nicole allow=all host=10.0.0.56 nat=yes and for configuration of my grandstream handytone 286 i got: sip server: 10.0.0.21 sip user id: christophe authenticate id: 4445 authenticate password: 4445 and as vocoder i got: G729 G729 G729 G729 PCMU PCMA PCMU sip registration: yes unregister at reboot: yes anyone know what i could be doing wrong ? -- Christophe De Coninck | Zarek K ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Christophe De Coninck | Zarek K http://www.zarekk.be mailto: [EMAIL PROTECTED] mailto: [EMAIL PROTECTED] attachment: banner.gif___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP phones
Michael Di Martino wrote: I am looking for a loud ringing SIP phone. I am presently using the Polycom and just cannot loud enough to hear it over the din in a collocation room. My Cisco 7960 has the loudest ring that I have ever heard, from any phone, period. -- Kristian Kielhofner ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New Channel Driver: chan_bluetooth
On Oct 20, 2004, at 5:37 AM, Theo Zourzouvillys wrote: after a couple of days work banging my head against the wall (bloody standards my arse), i've got chan_bluetooth to a point where it's starting to function - certianly more than just proof of concept now. Cool, I've been looking forward to something like this for months. I won't be able to play with it until the weekend, but it's great to see progress. Scott ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Personal Phone Gateway PCI and USB Phone.-
Hi, I was looking at a board that I have from tjnet.com and noticed that I looks almost the same as digiums X100p. I was wondering if anyone has tried using this board with Asterisk ? I also have a pair of USB phones form tjnet and I believe they use the tiger 560B and / or Tiger320. Their website is www.tjnet.com Has anyone played around with this hardware and made it work. Are there any linux drivers for them ? Greetings, Francisco ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] cannot call Grandstream
I also will get this message sometimes: -- Got SIP response 481 no such call back from 192.168.42.234 but I do have canreinvite=no in the appropriate section in sip.conf... On Wed, Oct 20, 2004 at 09:53:21AM -0400, Michael George wrote: I am having trouble with a Grandstream Budgetone 101. It's at firmware 1.0.5.10 and I'm running * 1.0.0. I have the phone getting a DHCP address and * expects it to register. When I reboot the phone it does register just fine. However, after a while * cannot contact the phone. I will call the phone and * will tell me: -- Called grandstream1 Oct 20 09:41:16 WARNING[98310]: chan_sip.c:681 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Critical Request) Looking in teh archives, it seems that that indicates that the registration is expired. I've got the phone set to 60m register intervals (and * acks that when the phone registers) but after the hour it doesn't re-register. I've also tried 15m and 2m register timeouts. I have Sip Registration and Unregister on Reboot both set to Yes on the phone. Register Expiration is 60. The phone is at 192.168.42.234 and * is as 192.168.1.3. Both internal but no NAT between them. And the initial registration works fine. I've searched through the mail list archives and tried all the suggestions I could find there, but the phone behaves the same: registration appears to be lost. Incidentally, I set the phone to a static IP (192.168.42.99) and also set * from host=dynamic to host=192.168.42.99 but * couldn't call the phone at all after that. (I did graceful restarts on * between the change). Can anyone see what I might be missing? I don't have the SIP UserID or Authenticate ID set to the phone's extension, but the SIP User ID is the same as the Authenticate ID which is the same as the context in *'s sip.conf. It doesn't seem that would have an effect, but I thought I'd mention it. Thanks! -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- [This E-mail scanned for viruses by Declude Virus] -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Wonderful Success with PAP2-NA
I have 2 of this PAP2, its a good product and it works with SER+Asterisk very well but it is not perfect, I can feel this humming noise on the line and I think it would be perfect if this device had more codecs like gsm or iLBX to better interaction of asterisk ivr voicemail and lower BW consumption when talk with x-lite, sjphone etc... Rafael On Wed, 20 Oct 2004 05:30:56 -0500, Your Own ISP .com [EMAIL PROTECTED] wrote: What about all the horrible reports of humming noise on the line on many of these units? Thanks, Todd Routhier Lightwave Technologies, LLC. -- Start Your Dialup Internet Service! http://www.YourOwnISP.com Lightwave Technologies, LLC. http://www.LightWaveTech.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Boehm Sent: Tuesday, October 19, 2004 3:37 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Wonderful Success with PAP2-NA Finally got authorized to purchase some PAP2-NA's from Linksys's. Works like a charm with Asterisk. Web configuration has TONS of options and looks nice. Able to put line1 and line2 on seperate asterisk servers. Hooked a 4-line vtech phone up to 2 PAP2-NAs and basically had created a 4 line ATA for $100. -Matthew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- rrgv ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] New Channel Driver: chan_bluetooth
I'll see that I can test this later this week or this weekend. -Original Message- From: Theo Zourzouvillys [mailto:[EMAIL PROTECTED] Sent: Wednesday, October 20, 2004 7:37 AM To: [EMAIL PROTECTED] Cc: [EMAIL PROTECTED] Subject: [Asterisk-Users] New Channel Driver: chan_bluetooth Hi, after a couple of days work banging my head against the wall (bloody standards my arse), i've got chan_bluetooth to a point where it's starting to function - certianly more than just proof of concept now. The code is far, far from stable, and only been tested with one dongly (Cambridge Silicon), and two devices (Nokia 6310i and a HBH-200 headset) - and even those devices are not FULLY functional yet. I can make and recieve calls through the devices, at times though ;) So why am I announcing it when it doesn't even work properly? I'd like to get people testing it with their devices and bluetooth adapters, see if it will pair, see debug output - see what phones/headsets do and don't work, see what documentation i need to add for setting it up, and of course get some general feedback about peoples wishes/views on the module. Code is at http://www.crazygreek.co.uk/chan_bluetooth Follow instructions on that page, or ask on list [1] when you encounter problems! Please, let me have some feedback so i can get this driver working properly with all HeadsetProfile devices! ~ Theo [1] - http://tribble.crazygreek.co.uk/cgi-bin/mailman/listinfo/chan _bluetooth -- Theo P. Zourzouvillys [EMAIL PROTECTED] http://www.crazygreek.co.uk/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Control access to external dialing
Wondering if anyone could give me a tip on controlling access under the following scenario. I have an ATA connected to a legacy pbx as a trunk line. I want to control who can make calls on this trunk. I cannot set restrictions on the users via the pbx, so I would like to be able to assign a passcode for people so they can dial out using this trunk line... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] DUNDi in stable? (New subject)
That's good to know. But, not to sound dumb, I'm not a heavy CVS user, how do I get the latest stable? As of now. The way I'm used to doing it is: export CVSROOT=:pserver:[EMAIL PROTECTED]:/usr/cvsroot cvs login cvs checkout zaptel libpri asterisk But that doesn't tell me if that's head or stable. The instructions say: cvs checkout -r v1-0 zaptel libpri asterisk asterisk-addons asterisk-sounds For stable. But my understanding is that will give me version 1.0; no bug fixes since the release of 1.0. I want the latest w/ bug fixes but no new features. My voicemail right now is not rigged for database support and such, just the standard voicemail.conf; So if I go to the latest, I don't want to be forced to retrofit my current voicemail setup. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kevin P. Fleming Sent: Tuesday, October 19, 2004 6:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] DUNDi in stable? (New subject) Deon Rodden wrote: When do you think the last stable CVS will be available before lots of stuff begins to change? I want to find the best possible Asterisk and stick with it, for some time, maybe until 2.0; If I get CVS right now, what if tomorrow or the day after he comes out with a better CVS. There is no need to rush and pull it now... you can always pull a snapshot of the tree as of any past date if you want, it's easy to do. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] grandstream handytone 286 problem
Christophe, Just for starters try changing your SIP user ID in the 286 to and 4445 and see if they register then. I have several 286's and they all work fine, but I don't use names, just numbers. Regards Dave -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of christophe de coninckSent: 20 October 2004 15:37To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] grandstream handytone 286 problem update:now I get this after i repowered the grandstream handytone 286:*CLI Oct 20 16:34:33 NOTICE[262160]: chan_sip.c:7532 handle_request: Registration from 'sip:[EMAIL PROTECTED];user=phone' failed for '10.0.0.55 On Wed, 2004-10-20 at 15:30, christophe de coninck wrote: hey,i got asterisk running with softphones but now I received a set of grandstream handytone 286's now if I run the setup and configure everything like supposed to be it doesn't work, i hear a ringing tone , after 30secs it hangs up and that's itin sip.conf i have:[4445]secret=4445type=friendusername=christopheallow=allhost=10.0.0.55nat=yes[]secret=type=friendusername=nicoleallow=allhost=10.0.0.56nat=yesand for configuration of my grandstream handytone 286 i got:sip server: 10.0.0.21sip user id: christopheauthenticate id: 4445authenticate password: 4445and as vocoder i got:G729G729G729G729PCMUPCMAPCMUsip registration: yesunregister at reboot: yesanyone know what i could be doing wrong ? -- Christophe De Coninck | Zarek K ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Christophe De Coninck | Zarek K http://www.zarekk.bemailto: [EMAIL PROTECTED] mailto: [EMAIL PROTECTED] banner.gif___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] DUNDi in stable? (New subject)
cvs checkout zaptel libpri asterisk == HEAD cvs checkout -r v1-0 zaptel libpri asterisk == STABLE with bug fixes. bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Deon Rodden Sent: Wednesday, October 20, 2004 9:58 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] DUNDi in stable? (New subject) That's good to know. But, not to sound dumb, I'm not a heavy CVS user, how do I get the latest stable? As of now. The way I'm used to doing it is: export CVSROOT=:pserver:[EMAIL PROTECTED]:/usr/cvsroot cvs login cvs checkout zaptel libpri asterisk But that doesn't tell me if that's head or stable. The instructions say: cvs checkout -r v1-0 zaptel libpri asterisk asterisk-addons asterisk- sounds For stable. But my understanding is that will give me version 1.0; no bug fixes since the release of 1.0. I want the latest w/ bug fixes but no new features. My voicemail right now is not rigged for database support and such, just the standard voicemail.conf; So if I go to the latest, I don't want to be forced to retrofit my current voicemail setup. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kevin P. Fleming Sent: Tuesday, October 19, 2004 6:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] DUNDi in stable? (New subject) Deon Rodden wrote: When do you think the last stable CVS will be available before lots of stuff begins to change? I want to find the best possible Asterisk and stick with it, for some time, maybe until 2.0; If I get CVS right now, what if tomorrow or the day after he comes out with a better CVS. There is no need to rush and pull it now... you can always pull a snapshot of the tree as of any past date if you want, it's easy to do. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: Asterisk-Users Digest, Vol 3, Issue 264
I've tried having only a single type=friend section for sip.broadvoice.com and it didn't make a difference. Having the two separate peer and user sections with the same name was something I got from one of the documentation examples. This is an excerpt from sip.conf with a single section for broadvoice that also doesn't seem to run the right context. register = 555111:[EMAIL PROTECTED] [sip.broadvoice.com] type=friend context=incoming host=sip.broadvoice.com nat=yes canreinvite=no dtmfmode=inband insecure=very username=555111 fromuser=555111 fromdomain=sip.broadvoice.com secret=password disallow=all allow=ulaw maxexpirey=15 -Original Message- Again, this is all speculation, but I've never seen two definitions for a user...maybe it doesn't know which to use, so it goes to general where the context is incoming1. Try changing the username for one of the sip.broadvoices... Message: 6 Date: Tue, 19 Oct 2004 14:52:59 -0400 From: Emerson, Michael [EMAIL PROTECTED] Subject: [Asterisk-Users] incorrect context called when receiving call on SIP channel To: '[EMAIL PROTECTED]' [EMAIL PROTECTED] Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii I am setting up asterisk to work with Broadvoice and so far am working with a very simple dialplan. My sip.conf file is below. My problem is I think the wrong context is being called on incoming calls. I think the [Incoming] context should be run and instead the [incoming1] context is used. Can anyone see what I am doing wrong? [general] disallow=all allow=ulaw port=5060 bindaddr=0.0.0.0 externip=1.2.3.4 localnet=192.168.1.20/255.255.255.0 context=incoming1 maxexpirey=180 defaultexpirey=160 canreinvite=no tos=reliability srvlookup=yes videosupport=no dtmfmode=inband nat=yes register = 555111:[EMAIL PROTECTED] [sip.broadvoice.com] type=peer context=incoming host=sip.broadvoice.com nat=yes canreinvite=no dtmfmode=inband insecure=very [sip.broadvoice.com] type=user username=555111 fromuser=555111 fromdomain=sip.broadvoice.com secret=password disallow=all allow=ulaw maxexpirey=15 host=sip.broadvoice.com nat=yes canreinvite=no dtmfmode=inband Mike Emerson Vital Basics Inc. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] DUNDi in stable? (New subject)
-Original Message- From: Deon Rodden [mailto:[EMAIL PROTECTED] Sent: Wednesday, October 20, 2004 10:58 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] DUNDi in stable? (New subject) That's good to know. But, not to sound dumb, I'm not a heavy CVS user, how do I get the latest stable? As of now. The way I'm used to doing it is: export CVSROOT=:pserver:[EMAIL PROTECTED]:/usr/cvsroot cvs login cvs checkout zaptel libpri asterisk But that doesn't tell me if that's head or stable. The instructions say: cvs checkout -r v1-0 zaptel libpri asterisk asterisk-addons asterisk-sounds For stable. But my understanding is that will give me version 1.0; no bug fixes since the release of 1.0. I want the latest w/ bug fixes but no new features. cvs checkout -r v1-0 will get you the latest for version 1.0 including bugfixes and anything else that is added to the 1.0 branch. Using cvs without the -r v1-0 gets you head. Good luck, Robert Jackson ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Wonderful Success with PAP2-NA
The humming noise comes from the Power units of these devices and has nothing to do with the quality of the SIP device. You can reduce this noise if your power unit has a longer cord or if you are using an expensive Power unit that has similar input and output. We found this problem on our netweb-301 / 302 phones wherein when we used the Australia (220 V) Only Power unit, in Sydney, it has humming noise. But when we use the 110-240 V Power units in USA, which are a little expensive, our phones work pindrop perfect. Hence we are supplying the 110-240V dual power units only on all our IP Phones sold now. Seshu http://ipphone.eezeephone.com 732-213-2422 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rafael J. Risco G.V. Sent: Wednesday, October 20, 2004 10:53 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Wonderful Success with PAP2-NA I have 2 of this PAP2, its a good product and it works with SER+Asterisk very well but it is not perfect, I can feel this humming noise on the line and I think it would be perfect if this device had more codecs like gsm or iLBX to better interaction of asterisk ivr voicemail and lower BW consumption when talk with x-lite, sjphone etc... NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Trunking Gateway with E100P
Hello I am asterisk's new user, and at present I meet working at the construction of a Trunking Media Gateway with a card E100P and that he be controlled with protocol MGCP. I wantto know if possible to accomplish that set-up, and how it would be possible to take end for it.I would thank that somebody that have experience with this theme giveme somehelp. Thanks Ricardo Barraza Sánchez Memorista Ing. Civil Electricista Universidad de Chile ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Control access to external dialing
Luke, I have a situation like yours, mine is to enable an IAX2 call between two servers and then break out to a trunk. All I have done is added a six digit code in front of the number (eg Birthdate ,210573 or 052173 if in US), and then stripted the six digits before dialing. You only tell the people you want to be able to dial out the six digit code. Regards Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Luke Catranis Sent: 20 October 2004 15:55 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Control access to external dialing Wondering if anyone could give me a tip on controlling access under the following scenario. I have an ATA connected to a legacy pbx as a trunk line. I want to control who can make calls on this trunk. I cannot set restrictions on the users via the pbx, so I would like to be able to assign a passcode for people so they can dial out using this trunk line... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] DUNDi in stable? (New subject)
www.bkw.org/dundi.tar.gz should compile and install on stable bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Brian West Sent: Wednesday, October 20, 2004 10:03 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] DUNDi in stable? (New subject) cvs checkout zaptel libpri asterisk == HEAD cvs checkout -r v1-0 zaptel libpri asterisk == STABLE with bug fixes. bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Deon Rodden Sent: Wednesday, October 20, 2004 9:58 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] DUNDi in stable? (New subject) That's good to know. But, not to sound dumb, I'm not a heavy CVS user, how do I get the latest stable? As of now. The way I'm used to doing it is: export CVSROOT=:pserver:[EMAIL PROTECTED]:/usr/cvsroot cvs login cvs checkout zaptel libpri asterisk But that doesn't tell me if that's head or stable. The instructions say: cvs checkout -r v1-0 zaptel libpri asterisk asterisk-addons asterisk- sounds For stable. But my understanding is that will give me version 1.0; no bug fixes since the release of 1.0. I want the latest w/ bug fixes but no new features. My voicemail right now is not rigged for database support and such, just the standard voicemail.conf; So if I go to the latest, I don't want to be forced to retrofit my current voicemail setup. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kevin P. Fleming Sent: Tuesday, October 19, 2004 6:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] DUNDi in stable? (New subject) Deon Rodden wrote: When do you think the last stable CVS will be available before lots of stuff begins to change? I want to find the best possible Asterisk and stick with it, for some time, maybe until 2.0; If I get CVS right now, what if tomorrow or the day after he comes out with a better CVS. There is no need to rush and pull it now... you can always pull a snapshot of the tree as of any past date if you want, it's easy to do. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] new here : logic of ser and asterisk all confused---longish
Hello,
comments inline...
--- Iqbal [EMAIL PROTECTED] wrote:
if (uri =~ sip:[EMAIL PROTECTED]){
log(1, Forwarding to Asterisk\n);
rewritehostport(193.218.160.25:5090);
break;
}
which I think means any number starting with a 2 send to asterisk
server..now when I dial this , in the SER logs it shows the message
Forwarding to Asterisk, and then waits, but in asterisk sip debug there
is nothing, not a sausage
I am afraid you are not sending the calls to Asterisk, but just rewriting the host and
port. After
rewriting, forward/relay the calls to Asterisk.
SERADDRESS=sip.ipclouds.co.uk:5060
[OUTGOING]
; Line below added for ser --- iqbal
exten = 1000,1,Dial(SIP/[EMAIL PROTECTED],20,r)
seeing all this it would seems that asterisk and ser go into a loop,
cause extensions simply sends it back to SER, which is what seems to
happen, and the ser.cfg sends it back to extensions.
Dont Asterisk complain about a '482 Loop Detected' error? The Dial statement will
create a new
INVITE and will be relayed to SER, which will send it back to Asterisk, thus resulting
in a loop.
Asterisk will drop this call. For dialing extensions use either Asterisk or SER. IMO,
use ser for
all extension dialing, and have appropriate forwarding and failure routing in the
ser.cfg to send
calls to Asterisk for the PBX features and voicemail.
Regards, Girish
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RE: [Asterisk-Users] DUNDi in stable? (New subject)
On Wed, 2004-10-20 at 10:58, Deon Rodden wrote: That's good to know. But, not to sound dumb, I'm not a heavy CVS user, how do I get the latest stable? As of now. The way I'm used to doing it is: export CVSROOT=:pserver:[EMAIL PROTECTED]:/usr/cvsroot cvs login cvs checkout zaptel libpri asterisk But that doesn't tell me if that's head or stable. The instructions say: cvs checkout -r v1-0 zaptel libpri asterisk asterisk-addons asterisk-sounds For stable. But my understanding is that will give me version 1.0; no bug fixes since the release of 1.0. I want the latest w/ bug fixes but no new features. My voicemail right now is not rigged for database support and such, just the standard voicemail.conf; So if I go to the latest, I don't want to be forced to retrofit my current voicemail setup. It will do exactly what you want it to. Because v1-0 is a *branch* tag CVS handles it a little differently from a regular tag. By checking out the branch you are now rooted on the stable 1.0 branch but you will still get the latest versions committed to that branch (read: you will get all the bug fixes). Runnning cvs update from your sandbox will contine to bring down patches applies to the 1.0 branch. BTW... if you want to convert your existing cvs HEAD over to the stable branch you can run update -r v1-0 instead of checkout. -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Control access to external dialing
On Wed, 2004-10-20 at 10:55 -0400, Luke Catranis wrote: Wondering if anyone could give me a tip on controlling access under the following scenario. I have an ATA connected to a legacy pbx as a trunk line. I want to control who can make calls on this trunk. I cannot set restrictions on the users via the pbx, so I would like to be able to assign a passcode for people so they can dial out using this trunk line... wiki or google DISA -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Control access to external dialing
That would work, but I have multiple people and I my customer needs to be able to track who is using the line and for what. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David J Carter Sent: Wednesday, October 20, 2004 11:37 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Control access to external dialing Luke, I have a situation like yours, mine is to enable an IAX2 call between two servers and then break out to a trunk. All I have done is added a six digit code in front of the number (eg Birthdate ,210573 or 052173 if in US), and then stripted the six digits before dialing. You only tell the people you want to be able to dial out the six digit code. Regards Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Luke Catranis Sent: 20 October 2004 15:55 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Control access to external dialing Wondering if anyone could give me a tip on controlling access under the following scenario. I have an ATA connected to a legacy pbx as a trunk line. I want to control who can make calls on this trunk. I cannot set restrictions on the users via the pbx, so I would like to be able to assign a passcode for people so they can dial out using this trunk line... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Vmail.cgi Bahhh!!
asterisk records the files to the filesystem with root permissions, which a properly configured apache installation doesn't have access too. Actually, it should only record the files with root permissions if asterisk itself is running as root. Which you shouldnt be doing in the first place, serious security problem if asterisk gets a few exploitable vulnerabilities. And even if you go about chmodding in a cron job, you shouldnt chmod it 777, it should at least be 770 with the same group as apache. Try running asterisk as a regular user, thats in the same group as apache. Then it should create the files so they are readable by apache, but retain write permissions for asterisk. -- Josh Krueger Urban Communications http://www.urbancom.net/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Wildcard X100P/India
Can anyone tell me if they have successfully deployed the X100P in India or any where in Southeast Asia? Thank you, Jonathan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Vmail.cgi Bahhh!!
Are there no permissions issues that will ever come up by running Asterisk as a non-root user? My Asterisk server is a dedicated/closed system, only I have access to ssh into it. It's also behind an external firewall that only allows certain udp ports through from the world. And ssh from my specific static IP. So I tried my best to keep the security tight. But if there's no performance impact or any permission downsides to running Asterisk as non-root, I'm game. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Josh Krueger Sent: Wednesday, October 20, 2004 11:35 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Vmail.cgi Bahhh!! asterisk records the files to the filesystem with root permissions, which a properly configured apache installation doesn't have access too. Actually, it should only record the files with root permissions if asterisk itself is running as root. Which you shouldnt be doing in the first place, serious security problem if asterisk gets a few exploitable vulnerabilities. And even if you go about chmodding in a cron job, you shouldnt chmod it 777, it should at least be 770 with the same group as apache. Try running asterisk as a regular user, thats in the same group as apache. Then it should create the files so they are readable by apache, but retain write permissions for asterisk. -- Josh Krueger Urban Communications http://www.urbancom.net/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] grandstream handytone 286 problem
Hey, The usage of numbers worked fine, everythings runs good now as far I know but there's only one problem. I got for each number a voicemailnumber eg number 4445 has number 44451 for voicemail, but then they have to enter a password. So i type 1235 (see example below from voicemail.conf) and it keeps saying login incorrect, anyone got any idea? works fine with kphone in linux. from voicemail.conf: 4445 = 1235,4445,[EMAIL PROTECTED] On Wed, 2004-10-20 at 17:28, David J Carter wrote: Christophe, Just for starters try changing your SIP user ID in the 286 to and 4445 and see if they register then. I have several 286's and they all work fine, but I don't use names, just numbers. Regards Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of christophe de coninck Sent: 20 October 2004 15:37 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] grandstream handytone 286 problem update: now I get this after i repowered the grandstream handytone 286 : *CLI Oct 20 16:34:33 NOTICE[262160]: chan_sip.c:7532 handle_request: Registration from 'sip:[EMAIL PROTECTED];user=phone' failed for '10.0.0.55 On Wed, 2004-10-20 at 15:30, christophe de coninck wrote: hey, i got asterisk running with softphones but now I received a set of grandstream handytone 286's now if I run the setup and configure everything like supposed to be it doesn't work, i hear a ringing tone , after 30secs it hangs up and that's it in sip.conf i have: [4445] secret=4445 type=friend username=christophe allow=all host=10.0.0.55 nat=yes [] secret= type=friend username=nicole allow=all host=10.0.0.56 nat=yes and for configuration of my grandstream handytone 286 i got: sip server: 10.0.0.21 sip user id: christophe authenticate id: 4445 authenticate password: 4445 and as vocoder i got: G729 G729 G729 G729 PCMU PCMA PCMU sip registration: yes unregister at reboot: yes anyone know what i could be doing wrong ? -- Christophe De Coninck | Zarek K ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Christophe De Coninck | Zarek K http://www.zarekk.be mailto: [EMAIL PROTECTED] mailto: [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Christophe De Coninck | Zarek K http://www.zarekk.be mailto: [EMAIL PROTECTED] mailto: [EMAIL PROTECTED] attachment: banner.gifbanner.gif___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Vmail.cgi Bahhh!!
Well, after messing with the voicemail folder and chmoding it, I could see my vmails. Woo Hoo.. I will figure out the permission issue I think, just needed to find the cause and this was it. Thanks a ton!! Thanks, Todd Routhier Lightwave Technologies, LLC. -- Start Your Dialup Internet Service! http://www.YourOwnISP.com Lightwave Technologies, LLC. http://www.LightWaveTech.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Josh Krueger Sent: Wednesday, October 20, 2004 10:35 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Vmail.cgi Bahhh!! asterisk records the files to the filesystem with root permissions, which a properly configured apache installation doesn't have access too. Actually, it should only record the files with root permissions if asterisk itself is running as root. Which you shouldnt be doing in the first place, serious security problem if asterisk gets a few exploitable vulnerabilities. And even if you go about chmodding in a cron job, you shouldnt chmod it 777, it should at least be 770 with the same group as apache. Try running asterisk as a regular user, thats in the same group as apache. Then it should create the files so they are readable by apache, but retain write permissions for asterisk. -- Josh Krueger Urban Communications http://www.urbancom.net/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Load Balaning on 2 E100P cards
I would like to know if it's possible to load balance calls on 2 E100P cards? In fact, I had an asterisk with a TE410P. 2 E1 are connected to the operator, and 2 others to an IVR PBX. Asterisk is used to place some calls in Voice over IP. I would like to know if it's possible, when I receive a call from my operator, if I can load balance it on my 2 others E1 connected to the PABX. I this case, if one PABX fail, I still had another one. show application congestion ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] manager interface to barge
Can the Manager interface be used to barge my phone into an existing conversation? db ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] meetme latency
I had a very unusual condition where an app_system call I was using was not completing for some reason before the call was put into the meetme. If I inserted a Wait(.5) before putting the call in the conferencethe meetme conference worked perfectly. Try inserting a Wait before dropping into the conference to see if this is a similar bug. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce Komito Sent: Wednesday, October 20, 2004 3:49 AM To: Bob Knight Cc: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] meetme latency For what it's worth, I have the same observation. Meetme used to work great, but sometime in the last few (3-4) months, it seems to have developed significant latency. Our echo test is also normal (way under a second), as are non-meetme calls. Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 On Tue, 19 Oct 2004, Bob Knight wrote: I am pretty sure that I had used meetme in the past (many months ago) with great results. Small number of users, mixed connections, IAX2 and SIP. For the past month or so, meetme has been a real pain due to very large latency. I can take 2 phones on the local lan and still get many seconds of latency. This makes it really hard to carry on a conversation. If I try to have folks join in over the net, we end up with 4 to 5 second latency. Is this normal, or do I have a problem. I am running 2.6.8ish kernel with no zap hardware. I am using the 2.6ish ztdummy. zttest looks ok. Echo test and phone calls are great. I think it is only when I get into the pseudo zap driver that I start having problems. Is it time for me to check out app_conference? -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This message has been categorized as Legitimate by Bayesian Analyzer. If you do not agree, please click on the link below to train the Analyzer. http://216.162.162.39/bt/a.aspx?M=C:%5Csmtpmail%5CBayesTraining%5C2004-1 0-19%5Cfdb007959f614e6190803a5c35248faeC=2 -- --- This message has been inspected by DynaComm i:mail --- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
