Re: [Asterisk-Users] Best ATA for general residential deployment??
[EMAIL PROTECTED] wrote: Time Warner provides an emta not an ATA and the technology is different. You do not even need internet connection for that and runs over their own private network through DOCSIS. Who manufacturers that unit? Have you found a way to interface it to a PBX? -- Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 UK 44.207.183.0271 Cell: 264-235-5670 Yahoo IM: [EMAIL PROTECTED] -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_capi-cm 0.6.4 random outgoing MSN problem
There are three possibilities to set the CallingPartyNumber (own number for
outgoing):
1) Set(CALLERID(number)=12345)
before Dial()
2) Dial(CAPI/contr1/12345:${EXTEN}/)
3) Dial(CAPI/contr1/${EXTEN}/d,...) and 'defaultcid=12345' in capi.conf
with this defaultcid you can set a number for each interface in capi.conf
and activate that by the /d option. This is useful when you combined more
than one interface into one group, but need to use a correct (and
different) number on dialout with e.g. 'g1', because the dialplan
doesn't know which interface will be used.
Armin
On Thu, 23 Feb 2006, Faris Raouf wrote:
Thanks for that Peter!
I think your message solved my problem: I set the master number to be in group
1 (group=1) in capi.conf and called Dial with CAPI/g1 and it worked perfectly.
However, with group=1 in capi.conf for the master number, at the moment no
matter what I do I'm getting the master number presented as the CLI. This is
fine by me because it is exactly what I want, but it is all very confusing :-)
Faris.
Peter Braidwood wrote:
I have ISDN2 and 10 MSN numbers and also have just upgraded to 1.2.4 and
chan_capi-cm and have this working completely perfectly
Capi.conf
[general]
nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8
language=en
[ISDN1]
isdnmode=msn
incomingmsn=*
controller=1
softdtmf=1
accountcode=
context=from-isdn
group=1
devices=2
bit of extensions.conf, I dial 9 for an outside line
[pstn]
exten = _9./321,1,Dial(CAPI/g1/01234567890:${EXTEN:1})
exten = _9./322,1,Dial(CAPI/g1/01234567891:${EXTEN:1})
exten = _9./323,1,Dial(CAPI/g1/01234567892:${EXTEN:1})
exten = _9./324,1,Dial(CAPI/g1/01234567893:${EXTEN:1})
exten = _9./326,1,Dial(CAPI/g1/01234567894:${EXTEN:1})
exten = _9./327,1,Dial(CAPI/g1/01234567895:${EXTEN:1})
exten = _9./328,1,Dial(CAPI/g1/01234567896:${EXTEN:1})
exten = _9./350,1,Dial(CAPI/g1/01234567897:${EXTEN:1})
exten = _9./351,1,Dial(CAPI/g1/01234567898:${EXTEN:1})
exten = _9./352,1,Dial(CAPI/g1/01234567899:${EXTEN:1})
So when extension 326 dials out the cli that is presented would be
01234567894
Contact me off list if you want any further help.
Peter Braidwood
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Faris
Raouf
Sent: 23 February 2006 13:24
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] chan_capi-cm 0.6.4 random outgoing MSN problem
I've having a big problem after having upgraded to 1.2.4 and chan_capi-cm
0.6.4
When making outgoing calls I don't seem to have any control over the CLI
that is presented to the called party -- it can be any one of the MSNs
allocated to the line, allocated on what seems to be a random basis.
This is on a BT Business Highway line (which is essentially an ISDN2e
line with two built-in analog ports), configured with 8MSNs alongside the
single the master digital telephone number for the line.
With a much older version of chan_capi-cm (0.5.4 I think) and Asterisk
1.0.9 it was always the master number that was presented, and that is
actually what I want.
Obviously the format of capi.conf has changed between these two versions
of chan_capi-cm, so maybe I'm doing something wrong. Any suggestions
would be appreciated.
Here's my capi.conf (actual numbers changed to protect the innocent!)
[general]
nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8
; ulaw=yes;set this, if you live in u-law world instead of
; a-law
[123456]
; Master number for line - i.e. number for line before MSNs were
allocated
; and which still works and still accepts incoming calls.
isdnmode=msn
msn=01234123456
; incomingmsn=*
incomingmsn=123456
controller=1
softdtmf=1
accountcode=
context=isdn-in
echosquelch=1
echocancel=yes
; echotail=64
; callgroup=1
; deflect=12345678
devices=2
[123457]
; first MSN
msn=01234123457
; incomingmsn=*
incomingmsn=123457
isdnmode=msn
controller=1
softdtmf=1
accountcode=
context=isdn-in
echosquelch=1
echocancel=yes
; echotail=64
; callgroup=1
; deflect=12345678
devices=2
{repeated for next 7 MSNs}
And in extensions.conf I have:
[globals]
ISDN1=CAPI/123456
[mysip]
; GET OUTSIDE LINE (ISDN1 - dial 9)
ignorepat = 9
exten = exten = _9.,1,Dial(${ISDN1}/${EXTEN:1}/b)
exten = _9.,2,Playback(busy)
exten = _9.,3,Hangup
*
I've tried using ISDN1=CAPI/contr1 but it makes no difference.
I've tried leaving out the isdnmode=msn but it makes no difference
I've tried entering 01234123456 as the msn= line on all of the msn
entries in capi.conf but it makes no difference either.
And now I'm out of ideas and any help would be appreciated.
Thanks,
Faris.
p.s. sorry if this
[Asterisk-Users] not consistent log from asterisk
Hello, I have 2 channels in iax.conf [iaxfwd] type=user callerid= Free World Dialup inkeys=freeworlddialup auth=rsa context=incoming qualify=yes [iaxfwd-outbound] type=peer host=iax2.fwdnet.net username=xx secret=*** auth=md5 The problem is: When I tell FWD to call me I have this output in my asterisk consol: Executing Dial(IAX2/iaxfwd-outbound-3, SIP/|PHONE_1|30) in new stack If I comment iaxfwd-outbound channel [iaxfwd-outbound], then the output is correct: Executing Dial(IAX2/192.246.69.186:4569-1, SIP/PHONE_1|30) in new stack. (192.246.69.186:4569 : this is from FWD) Thank's for any help a+ -- Bayrouni ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] UK X100P installation help
Thanks greatly for this. I will give it a go with these cards. I was trying to use Diva ones before. ISDN was by far my preferred choice, if I could get it to work... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tim Robinson Sent: 23 February 2006 21:26 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] UK X100P installation help Hi Paul - We gave up on analogue a long time ago in favour of ISDN. I have 3 ISDN cards in my Asterisk box. Billion ISDN BRI Cards cost me approx £15 each from komplett.co.uk and are perfect. You need to use the bri-stuffed version of Asterisk. If you still have the ISDNline I would recommend you give it another shot. You get none of the echo, caller ID and hangup detection problems with ISDN. It Just Works. (TM) Rgds Tim Robinson ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] UK X100P installation help
Paul -
Let me know when you have the cards and if you need any help. Main
thing is to ensure that you have each card on a seperate IRQ. this is
ESSENTIAL! Unless the bios is able to assign specific IRQs to specific
cards it might be a bit of a fiddle. For £15 you can't go far wrong though.
There are also some new drivers written as a seperate channel visdn
which I have not yet tried ('if it ain't broke, don't fix...' etc) which
might be more elegant as they apparently overcome some of the IRQ
issues. check the wiki for more details!
Rgds
Tim
Paul J. Smith wrote:
Thanks greatly for this. I will give it a go with these cards. I was trying
to use Diva ones before. ISDN was by far my preferred choice, if I could get
it to work...
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Tim Robinson
Sent: 23 February 2006 21:26
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] UK X100P installation help
Hi Paul -
We gave up on analogue a long time ago in favour of ISDN. I
have 3 ISDN
cards in my Asterisk box. Billion ISDN BRI Cards cost me approx £15
each from komplett.co.uk and are perfect. You need to use the
bri-stuffed version of Asterisk.
If you still have the ISDNline I would recommend you give it another
shot. You get none of the echo, caller ID and hangup detection
problems
with ISDN. It Just Works. (TM)
Rgds
Tim Robinson
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Re: [Asterisk-Users] spandsp debug or logging
Anton Krall wrote: Maybe this is a stupid question but how to you enable debubg or logging on spandsp? I see you can do that for RXFAX but when people tell you to enable debug on spandsp, do they mean this with rxfax or how do you do it with spandsp? You can do it writing: exten = s,1,rxfax(/fax/file/path|debug) or the same with txfax. The logs are then written to (default) /var/log/asterisk/full -- Best regards, Bartosz Piec ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] pickup problem on Asterisk 1.2.4
Solved the problem downgrading zaptel 1.2.4 to 1.2.3. Mantaining the same configurations now everything works fine. Regards, _fangi_ -Messaggio originale- Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Per conto di Francesco Angi Inviato: martedì 21 febbraio 2006 14.35 A: Asterisk Users Mailing List - Non-Commercial Discussion Oggetto: [Asterisk-Users] pickup problem on Asterisk 1.2.4 Hi everybody, I'm facing a strange problem after upgrading Asterisk from 1.0.9 to 1.2.4. Sometimes, when receiving an incoming call from pstn, although my sip phones ring correctly (I've got both softphones and hardware phones), noone can pick up the call. Asterisk CLI shows me that the phones are ringing, then nothing happens, so there's no problem _after_ someone picked up, simply Asterisk doesn't notice a phone picked up! Upgrading Asterisk I only did some changes to my dialplan, according to the upgrade page. My card is a TE110P, this is my zapata file: [channels] language=it context=default signalling=pri_cpe switchtype=euroisdn overlapdial=yes pridialplan = unknown prilocaldialplan = unknown nationalprefix = 0 internationalprefix = 00 echocancel=yes echotraining = 100 echocancelwhenbridged=yes immediate=no group=1 language=it musiconhold=default channel = 1-15,17-31 Thanks for help, _fangi_ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Anyone using the GSMgateway from CyberTelecom ?
Do you have any success receiving the caller id with your TDM400 FXO? I have the same problem when I connect the GSM gateway to a SPA3000 FXO line and thought this a Sipura's problem. On a phone connected to the GSM gateway I can see the callerid, but not on the Sipura's PSTN line ... this is no more and no less the same problem as I do have. It appears it's then not really the TDM400 FXO module. I have another option to test: I do have a similar ATA like the Sipura, but made by Grandstream. It's here at home; I will take it to the office tomorrow and see if it can read the caller id from the GSM gateway. Even my gsm unit does indeed pass the callerID when I connect it to a cheap, dead simple analog phone! BTW: Do you have a manual for the gateway? Thanks Aldo, No I do not have a manual and I don't believe such a thing exist. Actually, that GSM gateway is a Dock-N-Talk kind of thing with the exception that the handset is imbedded, so pretty much no need of a manual. Is your Grandstream a HT-488? If so you might be able to simulate the spa3000 case. Please, let me know what happened. Best regards, Benchev ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] can't dial some particular numbers (providers ?) with asterisk sip / zap channels
I have a strange problem when calling some numbers with asterisk, I get an hangup for busy condition even if the phone at the other end isn't busy. I can route the calls via SIP to another carrier and then I have a SIP code 486 or I can terminate them on digium cards (E1) and I have an Hangup code 17 I know for sure that one of the numbers is hosted by a different provider than the one that has the de-facto monopoly here, so maybe is a final-provider problem, even if I don't understand what kind of strange signalling can reach that provider from my asterisk, I don't see nothing unusual on the cli, is like any other call ended for a real busy condition. More weird is that with the SIP route the called phone rings once, than stops and I get the 486. What have I've already tried : Set(CALLERID(number)=[a real traditional phone number]) before the dial SetTransferCapability(SPEECH) as far as I know the route calls follow is : linksys pap --sip-- asterisk (1.2 or 1.0) --iax-- asterisk server (1.2) --zap-- ..?.. Hangup cause 17 linksys pap --sip-- asterisk (1.2 or 1.0) --iax-- asterisk server (1.2) --sip-- ..?.. - 486 Busy here (but the end phone ringed once) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: chan_capi-cm-0.6.4
On Thu, Feb 23, 2006 at 02:45:25PM +0100, Armin Schindler wrote: Hello Armin, hello List I'm trying to get chan_capi working with asterisk from debian stable (asterisk 1.0.7, the debian version number is 1:1.0.7.dfsg.1-2). I managed to get it compiled by providing my own version of ast_copy_string. Hmm, this should be handled automatically by the config script. Does Debian use a patched version of Asterisk? Probably, yes. There is a bug-report on sourceforge: http://sourceforge.net/tracker/index.php?func=detailaid=1435172group_id=140312atid=746578 I'm getting exactly the same error. Interesting is, that I receive an INFO_IND *before* the CONNECT_IND. This looks like an interesting variation of Austrian ISDN to me. Maybe it is a variation of the ISDN line, but the driver should fix that. Sending INFO_IND with a call-reference (PLCI) which is assigned by CONNECT_IND later, is just an error of the isdn driver. You mean, the capi part of misdn? Should I report a bug against mISDN? If you use mISDN, why don't you use chan_misdn? How reliable is that? Any experience? Ralf -- Ralf Schlatterbeck email: [EMAIL PROTECTED] FAX: +43/2243/26465/23 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk hints
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Garth van Sittert Sent: 24 February 2006 07:50 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk hints Hi Mike I have build 18 on the IP10's and I have tried call-limit=1 but it still doesn't work. Do the extension phones need to have any settings changed to enable this feature? I could be wrong but I think setting call-limit breaks hints in 1.2.x This is what finally forced me to get to grips with the GROUP() commands for limiting calls. Can't help much more than that though as we use Snom's with hints. HTH Alex Information contained in this e-mail and any attachments are intended for the use of the addressee only, and may contain confidential information of Ubiquity Software Corporation. All unauthorized use, disclosure or distribution is strictly prohibited. If you are not the addressee, please notify the sender immediately and destroy all copies of this email. Unless otherwise expressly agreed in writing signed by an officer of Ubiquity Software Corporation, nothing in this communication shall be deemed to be legally binding. Thank you. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Contact Center
Can the asterisk support a coaching function for the Supervisor to tap onto a call and coach the agent as she speaks to the customer without the customer hearing it.? Customer database management softward (or CRM) is this being included? Best regards Stephen ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: SV: [Asterisk-Users] Problems with voicemail
I checked the permitions and updated the ones with the wrong permissions. No it is reading the number of messages correct, but as soon as I press 1 to listen it stops again. So again, I checked the permissions on the messagefolder but it seemed ok. I see now that another person on this lista has the exact same problem. Kind regards Roger -Original Message- From: Dinesh Nair [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Thu, 23 Feb 2006 20:00:30 +0800 Subject: Re: SV: [Asterisk-Users] Problems with voicemail On 02/22/06 23:11 Roger Lewau said the following: Connected to Asterisk 1.2.4 currently running on ns2 (pid = 47562) Verbosity is at least 9 -- Remote UNIX connection -- Executing VoiceMailMain(SIP/asterisk-0946, @sip) in new stack -- Playing 'vm-login' (language 'se') -- Playing 'vm-password' (language 'se') -- Playing 'vm-youhave' (language 'se') == Spawn extension (sip, 990, 1) exited non-zero on 'SIP/asterisk-0946' it's borking when attempting to read numbers. is sounds/digits populated with adequate perms ? -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo=== ===+ | for a in past present future; do | | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +== ===+ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: chan_capi-cm-0.6.4
On Fri, 24 Feb 2006, Ralf Schlatterbeck wrote: On Thu, Feb 23, 2006 at 02:45:25PM +0100, Armin Schindler wrote: Hello Armin, hello List I'm trying to get chan_capi working with asterisk from debian stable (asterisk 1.0.7, the debian version number is 1:1.0.7.dfsg.1-2). I managed to get it compiled by providing my own version of ast_copy_string. Hmm, this should be handled automatically by the config script. Does Debian use a patched version of Asterisk? Probably, yes. There is a bug-report on sourceforge: http://sourceforge.net/tracker/index.php?func=detailaid=1435172group_id=140312atid=746578 I'm getting exactly the same error. Ah yes. Sorry I missed that. 0.6.4 does not check the existense of ast_copy_string(), CVS HEAD does. I will try to fix that soon. Interesting is, that I receive an INFO_IND *before* the CONNECT_IND. This looks like an interesting variation of Austrian ISDN to me. Maybe it is a variation of the ISDN line, but the driver should fix that. Sending INFO_IND with a call-reference (PLCI) which is assigned by CONNECT_IND later, is just an error of the isdn driver. You mean, the capi part of misdn? Should I report a bug against mISDN? Yes. Maybe it is already fixed in mISDN and you have an older version? If you use mISDN, why don't you use chan_misdn? How reliable is that? Any experience? I don't have any experience with mISDN. I just noticed the big work going on with mISDN and I assume that the support is good. Armin ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] a2billing without IVR
Hello list, Is there any way to use a2billing without the IVR for the sip/iax users. (authentication is done by the user id and pass as user registers with asterisk). I want to dial the destination number to the asterisk. for example: user dials, exten =_011.,1,DeadAGI(a2billing) system will connect the destination and bill them. but right now we need to enter the destinationfollowed by the IVR prompts which i dont want. Thanks in advanved if anybody can help me. best regards shaon ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] digium TE405P and intel motherboard
Well - a Sangoma Card won't bring you your money back. At least not immidiately. ;-) And a expensive highend echo cancelling card is not a good replacement for a relatively cheap TE405. So let's try to bring your existing investion to work. I presume you checked that your machine is working again if you remove the TE405. (otherwise: buy new mainboard ;-]) Did the server just don't boot your OS or is the machine dead (as in no BIOS activity etc.) Did you put the card in one of the 5V 32bit slots? Did you try to use the other 32bit slot? Did you make sure the crad is not broken? (i.E. tried it in an other machine) We use a few TE405 on Intel TorreyPines and they at least boot. Regards, Chris patty McHenry schrieb: The right direction is here: http://www.sangoma.com/datasheets/p_aft-104d-specs?PHPSESSID=82b00b2122ed47a4ac6f4f56487d740f Subject: [Asterisk-Users] digium TE405P and intel motherboard Hi, Can please someone help me. I have successfully installed Asteriskathome 2.5 on a server with a Intel Server Board SE7525RP2. May issue is after placing the TE405P in the server, it is not booting anymore. Has anyone in here have the same experience? Can someone please point me to the right direction. Thanks in advance, Leonimar ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Polycom IP 601 Buddy Watch doesn't work after Asterisk reload
Hi, I configured Buddy Watch function on my Polycom IP 601. It works well, until I make a reload of Asterisk. After reload, if I give the show hints command in Asterisk's CLI, it says that there are no watcher for the extensions that I configured. Before the reload in the CLI appears: -= Registered Asterisk Dial Plan Hints =- 3002 : SIP/3002 State:Idle Watchers 1 3006 : SIP/3006 State:Idle Watchers 1 3003 : SIP/3003 State:Unavailable Watchers 1 3001 : SIP/3001 State:Idle Watchers 1 3000 : SIP/3000 State:Idle Watchers 1 After the reload in the CLI appears: -= Registered Asterisk Dial Plan Hints =- 3002 : SIP/3002 State:Idle Watchers 0 3006 : SIP/3006 State:Idle Watchers 0 3003 : SIP/3003 State:Unavailable Watchers 0 3001 : SIP/3001 State:Idle Watchers 0 3000 : SIP/3000 State:Idle Watchers 0 Asterisk sends a SIP NOTIFY message in which the field Subscription-State is: terminated; reason=probation and the phone responds with a ACK. I have then to restart the phone to reactivate the Buddy Watch function. Is there anybody that can help me with this problem? Is it a problem of the PBX or a problem of the phone? Thanks in advance, regards, Marco. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: SV: [Asterisk-Users] Problems with voicemail
This probably has nothing to do with your problem, but I had a problem with similar symptoms, except asterisk was actually crashing whenever I tried to access voicemail. It would sometimes say some digits, but never got far (never got as far as the actual message). Problem turned out, crazily enough, to be having zaptel compiled with CONFIG_ZAPTEL_MMX. Commented that out, recompiled, worked fine. Uncommented again, recompiled, and it would crash every time I accessed voicemail. I'm running CentOS 4, with a 2.6 kernel, and did use the make linux26 command. Oh, and I did read the warning about compiling mmx with an AMD processor, but this server has an Intel Celeron in it, so it should have been ok. Oh well. Joseph Tanner On 2/24/06, Roger Lewau [EMAIL PROTECTED] wrote: I checked the permitions and updated the ones with the wrong permissions. No it is reading the number of messages correct, but as soon as I press 1 to listen it stops again. So again, I checked the permissions on the messagefolder but it seemed ok. I see now that another person on this lista has the exact same problem. Kind regards Roger -Original Message- From: Dinesh Nair [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Thu, 23 Feb 2006 20:00:30 +0800 Subject: Re: SV: [Asterisk-Users] Problems with voicemail On 02/22/06 23:11 Roger Lewau said the following: Connected to Asterisk 1.2.4 currently running on ns2 (pid = 47562) Verbosity is at least 9 -- Remote UNIX connection -- Executing VoiceMailMain(SIP/asterisk-0946, @sip) in new stack -- Playing 'vm-login' (language 'se') -- Playing 'vm-password' (language 'se') -- Playing 'vm-youhave' (language 'se') == Spawn extension (sip, 990, 1) exited non-zero on 'SIP/asterisk-0946' it's borking when attempting to read numbers. is sounds/digits populated with adequate perms ? -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo=== ===+ | for a in past present future; do | | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +== ===+ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: chan_capi-cm-0.6.4
On Fri, Feb 24, 2006 at 10:43:31AM +0100, Armin Schindler wrote: Interesting is, that I receive an INFO_IND *before* the CONNECT_IND. This looks like an interesting variation of Austrian ISDN to me. Maybe it is a variation of the ISDN line, but the driver should fix that. Sending INFO_IND with a call-reference (PLCI) which is assigned by CONNECT_IND later, is just an error of the isdn driver. You mean, the capi part of misdn? Should I report a bug against mISDN? Yes. Maybe it is already fixed in mISDN and you have an older version? Quite current. Checkout of mqueue branch from three days ago. Well I'll report a bug. If you use mISDN, why don't you use chan_misdn? How reliable is that? Any experience? I don't have any experience with mISDN. I just noticed the big work going on with mISDN and I assume that the support is good. I'll try this as a second route. I'd prefer chan_capi though, because I have used it for quite some time ... Thanks a lot for your help. Ralf -- Ralf Schlatterbeck email: [EMAIL PROTECTED] FAX: +43/2243/26465/23 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: a2billing without IVR
Asterisk Sales wrote: mailto:asterisk-users@lists.digium.com Hello list, Is there any way to use a2billing without the IVR for the sip/iax users. (authentication is done by the user id and pass as user registers with asterisk). I want to dial the destination number to the asterisk. for example: user dials, exten =_011.,1,DeadAGI(a2billing) system will connect the destination and bill them. but right now we need to enter the destination followed by the IVR prompts which i dont want. Thanks in advanved if anybody can help me. Yes, this is all configurable from /etc/asterisk/a2billing.conf If you set use_dnid=YES then a2billing will pick up the destination from the number the user dialled. Set the following to turn off the IVR stuff: ; Play the balance to the user after the authentication (values : yes - no) say_balance_after_auth=NO ; Play the balance to the user after the call (values : yes - no) say_balance_after_call=NO ; Play the time the user can call (values : yes - no) say_timetocall=NO Hope this helps. -- -Barry Flanagan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] What business IP phone to use
On Fri, 2006-02-24 at 10:54 +1100, David Ankers wrote: Are you sure those switch figures are right? 16ms delay in the switch path sounds a bit long. Cisco's mid-range switches like the 2950 have switching times measured in micro seconds. Then again a 2626 procurve is only around $700. I meant micro-seconds, yes - my apologies. The 26xx series are ok, but I had specifically the 4108 in mind when I said 'good experience'. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] What business IP phone to use
It must be microseconds that is being quoted, as even the 2626 that you mention lists a less than 13.3 microsecond latency. - Brad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Ankers Sent: Thursday, February 23, 2006 6:54 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] What business IP phone to use Are you sure those switch figures are right? 16ms delay in the switch path sounds a bit long. Cisco's mid-range switches like the 2950 have switching times measured in micro seconds. Then again a 2626 procurve is only around $700. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Conrad Wood Sent: Friday, 24 February 2006 7:50 AM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] What business IP phone to use Simple formula: 1. Total Revenue 2. % of revenue derived from phone usage 3. =Cost of downtime by using SoHo or consumer gear. It's not a question of if a SoHo or low cost device will screw up, it is a question of when. This is 23 years of experience talking. Where I work, the value of #3 above is $16 Cdn a *second*. We are below 500 employees, so we fall into the SMB segment. Sometimes I'm appalled by statements that a $700 switch or a $400 phone isn't worth it. Huh?? Maybe in Absolutely right! for something as critical as switches cabling I always recommend to spend real money. Don't ever try to save money any equipment that is required to operate the business. (Had very good experience with HP procurves over the last 10 years or so). There is no point buying netgear or other low-cost switches for a business ever. The cost saving of being able to pin-point a cabling/NIC/bandwidth problem down to the port on the switch easily and quickly is wonderful. Combined with SNMP and all the other goodies good switches come with, our clients save a lot of money by paying me less for my time ( d'oh ;-) ). The difference can also cause unnecessary delays and therefor echo in the path. For example, procurve switches typically have 13ms switching time, the high-end netgears about 21ms. As soon as you stack a couple of switches you are talking 26ms vs 42ms extra delay in the path! I see no reason however to spend $400 on a single phone though, because if a single phone breaks, it's not going to bring your business to a standstill, is it? (I guess unless you only have one in the first place ;-) ) conrad ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] mysql problems
On Fri, 2006-02-24 at 09:44 +0800, Ronald Wiplinger wrote: My database machine is broken and I have to use another one. I made somewhere mistake(s) and get now in the debug file: [Feb 24 09:05:24] DEBUG[32664]: MySQL RealTime: Query: SELECT * FROM sip_buddies WHERE name = '886' [Feb 24 09:05:24] DEBUG[32664]: MySQL RealTime: Query Failed because: Can't find file: './astconf/sip_buddies.frm' (errno: 13) first of, errno 13 is 'permission denied', so I guess your mysql database is running as a user who hasn't got permissions to the file. --- which makes it a question for the mysql mailing list. Anyways, on linux, you can use ps axu to find out as what user mysql runs as. Then change permissions/ownership on the files to match. Conrad ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How can I debug spandsp?
Hi, I'm trying to use the spandsp fax-back facility and despite I can send and receive faxes, this is not working fine. I would like to get a debug of what is going on. I am using the flag debug, but I don't know if txfax is generating any log information or where it is saving it. I just don't find anything. My line in extensions.conf is: exten = ,1,txfax(/home/victor/testfax.tif|debug) And from the fax machine I get the fax signal and 'receive error'. That's all. Could anybody tell me what to do to trace this, please? Thanks, Victor. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] What's with Indications/SetLanguage/Zaptel/RingBack ?
Good morning everybody,
Can someone explain to me the interconnection
between
thesefour things: indications.conf,
SetLanguage(), zaptel.conf
and ring-back ? If
there is any !! :- )
I am having this case where some users cannot hear
ring back
from a DeadAGI script and it seems to be
interconnected to these items.
These users are from the iaxfriends table,
they_can_ hear ring-back from
a Dial command directly in extensions.conf, but
_not_ from a DeadAGI
that performs the same Dial.
SIP users, directly defined in sip.conf, don't have
any problem.
Both dial the same IAX route.
At some point I had no indications.conf and Eric
Wieling suggested
to add it, which is what I did, and from there SIP
users in sip.conf started
to have the ring-back, but still, my users from the
iaxfriend table still
can't hear it.
I use asterisk 1.0.9
Should I add "language=br" in the iaxfriend source
code to make it work ?
I tried to add SetLanguage in extensions.conf but
without real success.
I included the concerned files here, if anybody
could give me a hint, it would
be really appreciated !
Thanks in advance,
Frederic
-- extensions.conf
---
Calling this one does not give me ring back from
the script:exten = _0XX32316200,1,DeadAGI(fred.agi)exten =
_0XX32316200,2,Hangup;Dialing this one directly gives me the ring
backexten =
_10XX32316200,1,Dial(IAX2/provider/559132316200,60);exten =
_10XX32316200,2,Hangup
--- fred.agi
---#!/usr/bin/perluse DBI;use
Asterisk::AGI;$AGI = new
Asterisk::AGI;$AGI-answer();$dialstr =
"IAX2/provider/559132316200|60";$res = $AGI-exec("DIAL
$dialstr"); zaptel.conf
---loadzone =
usdefaultzone=us
indications.conf
---[general]country=br[us]description
= United States / North Americaringcadance = 2000,4000dial =
350+440busy = 480+620/500,0/500ring =
440+480/2000,0/4000congestion = 480+620/250,0/250callwaiting
= 440/300,0/1dialrecall
=!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440record
= 1400/500,0/15000info =
!950/330,!1400/330,!1800/330,0[br]description =
Brazilringcadance = 1000,4000dial = 425busy =
425/250,0/250ring = 425/1000,0/4000congestion =
425/250,0/250,425/750,0/250callwaiting = 425/50,0/1000;
Dialrecall not used in Brazil standard (using UK standard)dialrecall =
350+440; Record tone is not used in Brazil, use busy
tonerecord = 425/250,0/250; Info not used in Brazil standard
(using UK standard)info =
950/330,1400/330,1800/330--
sip.conf:sip friends thathears
ring-back[general].language=en.[382762]type=friendusername=382762context=somethingsecret=secretnat=yescanreinvite=noqualify=nohost=dynamiclanguage=brincominglimit=1
iax.conf[general]language=en.;all
users are in iaxfriends and they don't hear ringback in deadagi but
;here it from Dial in extensions.conf
--- iaxfriends
table
mysql show columns from
iaxfriends;
+-+-+--+-+-+---+|
Field |
Type | Null | Key | Default | Extra
|+-+-+--+-+-+---+|
accountcode | varchar(20) |
| |
| ||
name | varchar(40)
| | PRI
|
| ||
secret | varchar(40) | YES
| |
| || context |
varchar(40) | YES |
|
| ||
ipaddr | varchar(20) | YES
| |
| ||
port |
int(6) | YES | |
0 |
|| regseconds | int(11) | YES
| | 0
|
|+-+-+--+-+-+---+
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Re: [Asterisk-Users] Re: Anyone using the GSMgateway from CyberTelecom ?
Benchev is believed to have said: Thanks Aldo, No I do not have a manual and I don't believe such a thing exist. Actually, that GSM gateway is a Dock-N-Talk kind of thing with the exception that the handset is imbedded, so pretty much no need of a manual. Is your Grandstream a HT-488? If so you might be able to simulate the spa3000 case. Please, let me know what happened. Best regards, Benchev Hello Benchev. the unit I have has also a serial port. So while it is really easy, a matter of plugging in cables, sim, power, to set up for receiving and making calls, I have no idea how to send and receive sms messages. Or what should/can be done with the serial port. I guess there must be a use for it, or one could save the effort to put one there. I do have a GS HT-488; but while in the office I was in such a hurry that I did no test. Sorry! I'll be back there next week; I'll let you know how the test will end. Best regards, Aldo ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk hints
I am using IP10s also It was working fine, but you needed to go into telnet mode, to activate the busy lamp, with the hint option ... moreover, if you wanted to pick up the phone call, then you needed also to add another telnet command to handle this pickup ! I know that swissvoice has now build 20, which allows all this through the web GUI interface ! Hope, this helps ! JM On 2/24/06, Alex Barnes [EMAIL PROTECTED] wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED]] On Behalf Of Garth van Sittert Sent: 24 February 2006 07:50 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk hints Hi Mike I have build 18 on the IP10's and I have tried call-limit=1 but it still doesn't work. Do the extension phones need to have any settings changed to enablethis feature?I could be wrong but I think setting call-limit breaks hints in 1.2.x This is what finally forced me to get to grips with the GROUP() commandsfor limiting calls.Can't help much more than that though as we use Snom's with hints.HTHAlexInformation contained in this e-mail and any attachments are intended for the use of the addressee only, and may contain confidential information of Ubiquity Software Corporation.All unauthorized use, disclosure or distribution is strictly prohibited.If you are not the addressee, please notify the sender immediately and destroy all copies of this email.Unless otherwise expressly agreed in writing signed by an officer of Ubiquity Software Corporation, nothing in this communication shall be deemed to be legally binding.Thank you. ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How can I debug spandsp?
Victor Alvarez wrote: I'm trying to use the spandsp fax-back facility and despite I can send and receive faxes, this is not working fine. I would like to get a debug of what is going on. I am using the flag debug, but I don't know if txfax is generating any log information or where it is saving it. I just don't find anything. By default it is in /var/log/asterisk/full file. -- Best regards, Bartosz Piec ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How can I debug spandsp?
Victor Alvarez wrote: is going on. I am using the flag debug, but I don't know if txfax is generating any log information or where it is saving it. I just don't find anything. My line in extensions.conf is: exten = ,1,txfax(/home/victor/testfax.tif|debug) Asterisk's debug facilities need to be enabled before you'll get debugging information. Doug ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Analyzer for Milliwatt
app_milliwatt is a nice tool for a quick check of the line quality. Anyway, hearing to that tone for more than a minute is painful. Does anyone know the opposite application, i.e. an application, that hears and listens for a 1000 Hz tone and displays the quality in any unit? If not, I'll think about developing one. No, but it sure would be nice to have some tools to diagnose quality issues. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Polycom IP 601 Buddy Watch problems
Here is the SIP transaction log. Caller called 7176 (Cisco 7960) from outside PSTN line, 7185(polycom 601, ip: 192.168.2.104) is the phone which monitors 7176.Reliably Transmitting (no NAT) to 192.168.2.104:5060:NOTIFY sip:[EMAIL PROTECTED] SIP/2.0Via: SIP/2.0/UDP 192.168.2.66:5060;branch=z9hG4bK567ba18bFrom: sip:[EMAIL PROTECTED];tag=as28665c79To: Wang sip:[EMAIL PROTECTED];tag=1B2B2C20-22A9C0D1Contact: sip:[EMAIL PROTECTED]Call-ID: [EMAIL PROTECTED]CSeq: 107 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: presenceContent-Type: application/xpidf+xmlSubscription-State: activeContent-Length: 349?xml version=1.0?!DOCTYPE presence PUBLIC -//IETF//DTD RFC XPIDF 1.0//EN xpidf.dtdpresencepresentity uri=sip:[EMAIL PROTECTED];method=SUBSCRIBE /atom id=7176address uri=sip:[EMAIL PROTECTED];user=ip priority=0.80status status=inuse /msnsubstatus substatus=onthephone //address/atom/presence-- SIP read from 192.168.2.104:5060:SIP/2.0 500 Internal Server ErrorVia: SIP/2.0/UDP 192.168.2.66:5060;branch=z9hG4bK567ba18bFrom: sip:[EMAIL PROTECTED];tag=as28665c79To: Wang sip:[EMAIL PROTECTED];tag=1B2B2C20-22A9C0D1CSeq: 107 NOTIFYCall-ID: [EMAIL PROTECTED]Contact: sip:[EMAIL PROTECTED]Event: presenceUser-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.4.0064Content-Length: 0 -- SIP read from 192.168.2.104:5060:SUBSCRIBE sip:[EMAIL PROTECTED] SIP/2.0Via: SIP/2.0/UDP 192.168.2.104;branch=z9hG4bKfdb7ef6c9DE13403From: Wang sip:[EMAIL PROTECTED];tag=1B2B2C20-22A9C0D1To: sip:[EMAIL PROTECTED];tag=as28665c79CSeq: 29 SUBSCRIBECall-ID: [EMAIL PROTECTED]Contact: sip:[EMAIL PROTECTED]Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFEREvent: presenceUser-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.4.0064Authorization: Digest username=7185, realm=asterisk, nonce=4eb67954, uri=sip:[EMAIL PROTECTED], response=3d264007cfea7ea28cf53fd4f9b12417, algorithm=MD5Max-Forwards: 70Expires: 3600Content-Length: 0Transmitting (no NAT) to 192.168.2.104:5060:SIP/2.0 200 OKVia: SIP/2.0/UDP 192.168.2.104;branch=z9hG4bKfdb7ef6c9DE13403;received=192.168.2.104From: Wang sip:[EMAIL PROTECTED];tag=1B2B2C20-22A9C0D1To: sip:[EMAIL PROTECTED];tag=as28665c79Call-ID: [EMAIL PROTECTED]CSeq: 29 SUBSCRIBEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYExpires: 3600Contact: sip:[EMAIL PROTECTED];expires=3600Content-Length: 0---Reliably Transmitting (no NAT) to 192.168.2.104:5060:NOTIFY sip:[EMAIL PROTECTED] SIP/2.0Via: SIP/2.0/UDP 192.168.2.66:5060;branch=z9hG4bK7601a9bdFrom: sip:[EMAIL PROTECTED];tag=as28665c79To: Wang sip:[EMAIL PROTECTED];tag=1B2B2C20-22A9C0D1Contact: sip:[EMAIL PROTECTED]Call-ID: [EMAIL PROTECTED]CSeq: 108 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: presenceContent-Type: application/xpidf+xmlSubscription-State: activeContent-Length: 349?xml version=1.0?!DOCTYPE presence PUBLIC -//IETF//DTD RFC XPIDF 1.0//EN xpidf.dtdpresencepresentity uri=sip:[EMAIL PROTECTED];method=SUBSCRIBE /atom id=7176address uri=sip:[EMAIL PROTECTED];user=ip priority=0.80status status=inuse /msnsubstatus substatus=onthephone //address/atom/presence-- SIP read from 192.168.2.104:5060:SIP/2.0 500 Internal Server ErrorVia: SIP/2.0/UDP 192.168.2.66:5060;branch=z9hG4bK7601a9bdFrom: sip:[EMAIL PROTECTED];tag=as28665c79To: Wang sip:[EMAIL PROTECTED];tag=1B2B2C20-22A9C0D1CSeq: 108 NOTIFYCall-ID: [EMAIL PROTECTED]Contact: sip:[EMAIL PROTECTED]Event: presenceUser-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.4.0064Content-Length: 0Reliably Transmitting (no NAT) to 192.168.2.104:5060:NOTIFY sip:[EMAIL PROTECTED] SIP/2.0Via: SIP/2.0/UDP 192.168.2.66:5060;branch=z9hG4bK212f2520From: sip:[EMAIL PROTECTED];tag=as28665c79To: Wang sip:[EMAIL PROTECTED];tag=1B2B2C20-22A9C0D1Contact: sip:[EMAIL PROTECTED]Call-ID: [EMAIL PROTECTED]CSeq: 109 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: presenceContent-Type: application/xpidf+xmlSubscription-State: activeContent-Length: 344?xml version=1.0?!DOCTYPE presence PUBLIC -//IETF//DTD RFC XPIDF 1.0//EN xpidf.dtdpresencepresentity uri=sip:[EMAIL PROTECTED];method=SUBSCRIBE /atom id=7176address uri=sip:[EMAIL PROTECTED];user=ip priority=0.80status status=open /msnsubstatus substatus=online //address/atom/presence-- SIP read from 192.168.2.104:5060:SIP/2.0 500 Internal Server ErrorVia: SIP/2.0/UDP 192.168.2.66:5060;branch=z9hG4bK212f2520From: sip:[EMAIL PROTECTED];tag=as28665c79To: Wang sip:[EMAIL PROTECTED];tag=1B2B2C20-22A9C0D1CSeq: 109 NOTIFYCall-ID: [EMAIL PROTECTED]Contact: sip:[EMAIL PROTECTED]Event: presenceUser-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.4.0064Content-Length: 0 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk configuration for h323 calls
Hello,I'm new to Asterisk. I want to do the folloing job.I want to run Asterisk as a voip gateway to forward h323 calls to another gateway. my-gateway - Asterisk -- your-gateway h323 h323Is it possible to do this? If so, can anyone give me an idea how to do it? How many configuration files relates to this job? Can you give a sample configuration? Thank yo u in advance.Roda Yahoo! Autos. Looking for a sweet ride? Get pricing, reviews, & more on new and used cars.___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: Asterisk 1.2.4 IAX2 New Jitterbuffer Tuning
I was using IAX2 with ILBC and no trunking. I also set the resyncthreshold=-1 to turn it off. Still had major jitter problems. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: Thursday, February 23, 2006 6:44 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Re: Asterisk 1.2.4 IAX2 New Jitterbuffer Tuning After 2 weeks of messing around with every conceivable IAX2 and jitterbuffer configuration, I switched to SIP yesterday. Complaints went from 10-20 per day to ZERO. Literally overnight. I wonder if this is an ILBC frame size issue of some sort? Seems odd. I've got to add my name to the list here. We're just using GSM over our IAX links, and our jitterbuffer values look like this: maxjitterbuffer=1000 resyncthreshold=1000 maxjitterinterps=10 For the most part the new jitterbuffer actually yields much better quality than the old jitterbuffer, but when the resyncs happen, it's like the call has a lot of trouble getting get back on track. It flounders for quite a while, with badly broken audio, sometimes up to 20 seconds before coming back. I've tried hanging up as soon as event starts happening and then immediately calling the same number, and the channel comes back with crystal clarity. So it seems to me like there is something askew with the resync. If memory serves correctly, I believe I remember Mark applying a fix to the iax jitterbuffer and that fix had something to do with a counter rollover or something like that. That fix happened in the last week or so. I'm not sure if that would have been included in v1.2.4 or not, but might be worth a little research. I also opened a bug a month or two ago involving ilbc and iax, and someone else confirmed it was a bug. Don't have the bug number handy, but the problem related to a combination of iax trunking, jitterbuffer and ilbc. Disabling one of those consistently bypassed the problem. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Analyzer for Milliwatt
Rich Adamson wrote: app_milliwatt is a nice tool for a quick check of the line quality. Anyway, hearing to that tone for more than a minute is painful. Does anyone know the opposite application, i.e. an application, that hears and listens for a 1000 Hz tone and displays the quality in any unit? If not, I'll think about developing one. No, but it sure would be nice to have some tools to diagnose quality issues. Maybe the first approach should be to setup a test extension for recording the tone. The idea is to get best resolution possible in real time. Then process it as much as needed to get the info you want. Such an approach would give you more flexibility. For example, you could automatically place periodic test calls to various servers and have the recordings then forwarded to one server for analysis. That would minimize the impact on production asterisk servers. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Which Quad Port FXO is Best?
John Kelly wrote: I'm looking to handle 3 PSTN lines with my Asterisk server. I have a Digium TDM22B and Sipura 3000. The Sipura works great, but the TDM22B seems to have terrible problems with my board---virtually all peripherals need to be disabled in BIOS, and then there is terrible noise, terrible silence and virtually no ability to use DTMF in IVRs. I really wish the TDM22B worked, because I much prefer storing all my configurations on one device, and not needing separate peer accounts for each PSTN line. However, I don't have the skills or spare hardware to debug this quickly, and I'm really wanting to get on with the task of developing some AGI apps. I see several 4 port FXO Analog/SIP gateways on voipsupply.com: [$350] Clipcom 410: http://www.voipsupply.com/product_info.php?products_id=240 [$635] Mediatrix 1204: http://www.voipsupply.com/product_info.php?products_id=171 [$560] Patton 4114: http://www.voipsupply.com/product_info.php?products_id=863 I know it would be cheaper to buy two more Sipuras, but it might be worth the extra $$ to cut down on the power adapters and have a centralized point of administration, especially if it didn't involve dozens of browser mouse clicks to 3 separate HTTP servers. Reliability is the primary criterium, though. Can anybody give any recommendations? And are these digium problems unusual? We now have three installs of the Sangoma A200 (with echo can) and love them. They sound quality is very good and, so far, they have totally eliminated echo. Mike Clark ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk hints
Hi Jean-Marc I tried removing the call-limit setting. It still doesn't work. I am using a SNOM 360 to monitor the line status'. Do I still need to activate the busy lamp on the IP10S' or is this only if you want the IP10S' to monitor the hints? Garth Jean-Marc Salsa wrote: I am using IP10s also It was working fine, but you needed to go into telnet mode, to activate the busy lamp, with the hint option ... moreover, if you wanted to pick up the phone call, then you needed also to add another telnet command to handle this pickup ! I know that swissvoice has now build 20, which allows all this through the web GUI interface ! Hope, this helps ! JM On 2/24/06, *Alex Barnes* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: -Original Message- From: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] [mailto:asterisk-users- mailto:asterisk-users- [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]] On Behalf Of Garth van Sittert Sent: 24 February 2006 07:50 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk hints Hi Mike I have build 18 on the IP10's and I have tried call-limit=1 but it still doesn't work. Do the extension phones need to have any settings changed to enable this feature? I could be wrong but I think setting call-limit breaks hints in 1.2.x This is what finally forced me to get to grips with the GROUP() commands for limiting calls. Can't help much more than that though as we use Snom's with hints. HTH Alex Information contained in this e-mail and any attachments are intended for the use of the addressee only, and may contain confidential information of Ubiquity Software Corporation. All unauthorized use, disclosure or distribution is strictly prohibited. If you are not the addressee, please notify the sender immediately and destroy all copies of this email. Unless otherwise expressly agreed in writing signed by an officer of Ubiquity Software Corporation, nothing in this communication shall be deemed to be legally binding. Thank you. ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Important: Application DIALPLAN STANDARD/GUIDELINES needs to be established.
Hello Asterisk community. We have a small User-group in Melbourne Australia. Recently I brought up the issue of STANDARDS for dialing Applications on a PBX. This generated some interest but also the fact little has been done on this topic. Below is a rundown of our THREAD. (start from bottom and go up) I myself, feel this to be an important issue. With Asterisk being so programmable, anything can be done. But should it. I would like to see some type of guide line or standard for extension layouts. We have not been able to find any reference to this. However, I hope the greater Asterisk community has, and if so, please share. Thanks, James Well, it comes down to personal preference I think, we use *1 for VM, and check CLID to take a caller directly to their VM box if it exists, vairous other internal functions from *1-9, other externally accessible functions from *10-19, conference rooms *20-39, etc... We've had no problems, but then we run a controlled set of end-user hardware. I suppose for a rollout with unkown/mixed hardware some research is required to determine the reserved functions. So, yes, two ideas might be to have a prefix (that is ensured never to be used in real number space!) for all functions, the other would be to have a number to dial that drops the caller into a context containing all features, possible even with voice prompts... Just idle thoughts... James Gardiner wrote: Hello all, Well, I would like to bring note to this topic as an important issue. I am working on a AMP like application and want to standardize on number sequences. *MAIL and *PARK sound like good ideas, however, they are long button sequences. Using * for applications, I feel, looks a bit shaky as its well used with no formula by many companies for DND and other things. So for example. *PARK is *7275. I am pretty sure *72 is some type of feature on Cisco/sipura handsets so, the handset will upset these sequence of numbers. I was looking at bringing it all to a standard or 1 application number Park 17 VM 15 direct 152000 for extension 2000. 15*2000 direct to voicemail for 2000 Listen to MOH 1100 Test dial in context 1000 Etc. (There are many other options to consider.) Something like this; Could the group members please make comment on what each of them sees advantages and disadvantages of this idea. Or any better ones. I am really open to suggestions. I really need to solidify the dial plan and manual. Thanks, James -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of jurgen Sent: Wednesday, 22 February 2006 11:09 AM To: Melbn VOIP Subject: Re: [Melbn-VOIP] Standards for Dialing applications When I was making some dial plan decisions several months ago, I didn't see any real standards either, aside from some that telcos have used (*69 for recent calls, etc). So I just went and made up my own, based on words: *MAIL (*6245), *PARK, etc etc. They're easy for users to remember, and as long as the phones have letters on them as well as numbers, they're easy to dial. On 22/02/2006, at 9:59 AM, James Gardiner wrote: New Topic.. I am looking at writing some documentation for and users and also implementing different features in an Asterisk system. I have been looking around at different systems. Now the *NN appears to be common between manufactures. Is there a documented standard for this? Do they just make it up as they want? For example. There does not appear to be a standard for dialling Voicemail. Parking etc. I suppose, the simple question is. Is there one? If not, what is the consensus on dial codes for these options? For example what do well known vendors use. (Like cisco, etc) Thanks, James ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] mpg123 alternative?
Is there any particular reason for the native file format stuff to be in asterisk-addons as opposed to that code being merged into trunk? It isn't. You are mis-interpreting the information in this thread (it's been unclearly stated anyway). The only portion that is in asterisk-addons is format_mp3, which allows Asterisk to natively open MP3 files. However, that is of little use, when you can use sox to convert those files into slinear/ulaw/alaw/gsm/etc. so that no transcoding is needed when the audio is played to a caller. Right on... I did mis-interpret it without a doubt. Part of that was oriented around the thought that I was going to have to convert all files (not just mp3's), and that obviously would be a very high maintenance item over time. I read stuff into it that I shouldn't have, and jumped to the wrong conclusion. More coffee, please! ;) One of the reasons for wanting to address mp3's (natively) is that some of our bank customers subscribe to an annual service from another firm to provide them with professional message on hold services. If those messages where provided to the bank in mp3 format, I was looking for a relatively simply way for non-technical * users to copy the file and play it without the need for external players, conversion steps, etc. Scripting the copy process and doing the sox conversion will accomplish the same goal. Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] What business IP phone to use
Aha, micro seconds in networking terms is normally written usecs or us (actually it's the greek letter mu as in ulaw) rather than ms which are milliseconds seconds - what had me puzzled was that it was stated that this could harm the voice path! The difference can also cause unnecessary delays and therefor echo in the path. For example, procurve switches typically have 13ms switching time, the high-end netgears about 21ms. As soon as you stack a couple of switches you are talking 26ms vs 42ms extra delay in the path! There is then only 8 usecs between the two switches, how on earth would this make any difference to the voice path at all? Let alone induce any echo... Obviously the originally poster didn't understand the difference. And based on this, he's probably advising people not to use Netgear switches for voice, oh dear. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Watkins, Bradley Sent: Friday, 24 February 2006 10:08 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] What business IP phone to use It must be microseconds that is being quoted, as even the 2626 that you mention lists a less than 13.3 microsecond latency. - Brad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Ankers Sent: Thursday, February 23, 2006 6:54 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] What business IP phone to use Are you sure those switch figures are right? 16ms delay in the switch path sounds a bit long. Cisco's mid-range switches like the 2950 have switching times measured in micro seconds. Then again a 2626 procurve is only around $700. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Conrad Wood Sent: Friday, 24 February 2006 7:50 AM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] What business IP phone to use Simple formula: 1. Total Revenue 2. % of revenue derived from phone usage 3. =Cost of downtime by using SoHo or consumer gear. It's not a question of if a SoHo or low cost device will screw up, it is a question of when. This is 23 years of experience talking. Where I work, the value of #3 above is $16 Cdn a *second*. We are below 500 employees, so we fall into the SMB segment. Sometimes I'm appalled by statements that a $700 switch or a $400 phone isn't worth it. Huh?? Maybe in Absolutely right! for something as critical as switches cabling I always recommend to spend real money. Don't ever try to save money any equipment that is required to operate the business. (Had very good experience with HP procurves over the last 10 years or so). There is no point buying netgear or other low-cost switches for a business ever. The cost saving of being able to pin-point a cabling/NIC/bandwidth problem down to the port on the switch easily and quickly is wonderful. Combined with SNMP and all the other goodies good switches come with, our clients save a lot of money by paying me less for my time ( d'oh ;-) ). The difference can also cause unnecessary delays and therefor echo in the path. For example, procurve switches typically have 13ms switching time, the high-end netgears about 21ms. As soon as you stack a couple of switches you are talking 26ms vs 42ms extra delay in the path! I see no reason however to spend $400 on a single phone though, because if a single phone breaks, it's not going to bring your business to a standstill, is it? (I guess unless you only have one in the first place ;-) ) conrad ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GPS-enabled cell phone/PDA
I would like to capture the lat/lon coordinates from a GPS-enabled cell phone or PDA. Is this possible? Must I subscribe to this information from the cellphone network provider, or can I capture it without charge? What devices will broadcast the coordinates? Is there a device that will broadcast its position inband that can be captured by Asterisk? Can an SMS message include coordinates? The subject will willingly carry the device and will be aware that his location is being monitored, so privacy rights are not an issue. The subject will make periodic calls to the Asterisk server in order to record his movements. Does anyone have experience in this area? Its my understanding the cell phone coordinates are sent to the cell phone provider and their equipment reads (and holds) that data. Its not part of any data available to you in any form unless you talk to the cell provider and convience them you have a valid need. Highly unlikely in the US anyway. Even if you could convience them to provide it, they would likely demaand some sort of out-of-band data transmission facility. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] lspci don't have Tiger Jet Network Inc
hello all, do i must must see Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface when in install TDM2424E card i think so but i can not see this in lspci is there is software tools to make sure that my motherboard have pci 2.2 and see TDM2424E see it right b4install zaptel driver and did this kind of card needextra work than other TDM card Yahoo! Autos. Looking for a sweet ride? Get pricing, reviews, & more on new and used cars.___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom IP 601 Buddy Watch doesn't work after Asterisk reload
On 2/24/06, Marco Maiolini [EMAIL PROTECTED] wrote: Hi, I configured Buddy Watch function on my Polycom IP 601. It works well, until I make a reload of Asterisk. After reload, if I give the show hints command in Asterisk's CLI, it says that there are no watcher for the extensions that I configured. Before the reload in the CLI appears: -= Registered Asterisk Dial Plan Hints =- 3002 : SIP/3002 State:Idle Watchers 1 3006 : SIP/3006 State:Idle Watchers 1 3003 : SIP/3003 State:Unavailable Watchers 1 3001 : SIP/3001 State:Idle Watchers 1 3000 : SIP/3000 State:Idle Watchers 1 After the reload in the CLI appears: -= Registered Asterisk Dial Plan Hints =- 3002 : SIP/3002 State:Idle Watchers 0 3006 : SIP/3006 State:Idle Watchers 0 3003 : SIP/3003 State:Unavailable Watchers 0 3001 : SIP/3001 State:Idle Watchers 0 3000 : SIP/3000 State:Idle Watchers 0 Asterisk sends a SIP NOTIFY message in which the field Subscription-State is: terminated; reason=probation and the phone responds with a ACK. I have then to restart the phone to reactivate the Buddy Watch function. Is there anybody that can help me with this problem? Is it a problem of the PBX or a problem of the phone? It is a phone issue as the phone is supposed to try and resubscribe after 60 seconds which is an attribute in that message, but it doesn't. However, bug 6047 in Mantis has some code to try and provide a workaround for this issue. Testing would be greatly appreciated. -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Analyzer for Milliwatt
On Friday 24 February 2006 07:56, Paul wrote: Maybe the first approach should be to setup a test extension for recording the tone. The idea is to get best resolution possible in real time. Then process it as much as needed to get the info you want. Such an approach would give you more flexibility. For example, you could automatically place periodic test calls to various servers and have the recordings then forwarded to one server for analysis. That would minimize the impact on production asterisk servers. What is being discussed here is basically what I was planning on doing for an automatic VOIP quality check. Using miliwatt and analyzing it for pop/jitter/etc as well as sending other known waveforms and comparing what was received to what was expected and coming up with some quality number which would be fed back to the dialplan to adjust the least-cost routing paths. Essentially come up with a least cost but still good quality routing. :-) I've done absolutely nothing other than a little research and a lot of thinking about how to do it though. I did some research on digital click/pop removal for records as a way to detect poor quality, and then also some monkeying around with coppice's excellent DSP routines in spandsp. -A. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] fax receive using TDM400P
Ive been testing how to receive faxes using TDM400P cards and so far, after playing with gains, echocancell and echotraining on zapata.conf.. Ive ha dno luck, faxes come in as garbage or broken or with blank lines. Anybody has successfully done this? Any tips.. Also I have some ideas: 1. Is it really possible to get fxes on a fax machine using ATAs like the sipura 2002? Even using ulaw and pass-thru, is it possible? 2. Since the faxes is coming from PSTN thru the card, I guess asterisk will always stay in the middle right? No way around this. 3. Im also trying to receive faxes usign a TE110P card with spandsp, unicall and E1 R2MFC, no luck also, some stuff, garbage and broken faxes. Anybody done this sucessfuly? Hope anybody can share their thoughts and insight on this. Using the TDM400 card for any form of fax'ing (or modem use) is well known to be unreliable and, in most cases, totally unusable. The issue has been discussed many times over the last two years or so. There are no known workarounds. Its my understanding that lots of folks have spandsp working via T1 and/or PRI interfaces. The issues associated with the TDM400 card do not apply to the T1 cards. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem with T1 installation
Hi All,
Ihave installed a Digium TE110P card on an
Asterisk 1.2.1 system. Its connected directly to the PSTN. But I am unable to
make outbound calls on the zap channels. The light on the card is green.
Asterisk CLI shows all 24 channels when I give the command 'zap show channels'.
I also noticed that Asterisk CLI shows an incoming call every few seconds on the
24th channel. This must be some kind of a timing signal. This is he first time I
am configuring a T1 so I must have done something wrong I guess.
These are the commands I used to load the zap
module:
modprobe zaptel
modprobe wcte11xp
ztcfg -vvv
---
my zaptel.conf is as
follows:
span=1,1,0,esf,b8zsem=1-24loadzone =
usdefaultzone=us
--
the zapata.conf is as
follows:
[trunkgroups][channels]
group=1language=ensignalling=em_wusecallerid=yescallerid=asreceivedcontext=defaultechocancel=64echocancelwhenbridged=yesrxgain=1.0txgain=1.0channel
=
1-2group=2language=ensignalling=em_wusecallerid=yescallerid=asreceivedcontext=defaultechocancel=64echocancelwhenbridged=yesrxgain=1.0txgain=1.0channel
= 3-24
--
In extensions.conf i have
specified the following line:
[default]
exten =
_ZX,1,Dial(zap/g1/${EXTEN},15,tr)
--
When I try to dialusing the T1 lineI
get the following error :
Feb 24 06:56:53 NOTICE[5724]:
app_dial.c:1010 dial_exec_full: Unable to create channel of type 'Zap' (cause 0
- Unknown) == Everyone is busy/congested at this time
(1:0/0/1) == Auto fallthrough, channel 'SIP/7180-a103' status is
'CHANUNAVAIL'
Any ideas guys?
Thanks and regards,
Nitin
Joshi.
___
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Re: [Asterisk-Users] OT: VoIP over bonded link
On Thursday 23 Feb 2006 20:34, Colin Anderson wrote: It's stupid. Don't ever connect 2 different building with copper. Just wait until you get some kind of lightening hit or electrical fault, but make sure you are no where near it. Use fibre. Thanks for the reply. Unfortunately, the conduit for the provisioning of the new building is unsuitable for fibre (too many sharp bends) and we can't core out the concrete and put in a new conduit because of obstacles in the way that make laying new conduit impractical, so we are stuck with (existing) copper. We already have copper-to-copper connections of different types (electrical, security etc) between the buildings so a lightning strike is going to hose us no matter what. In that case, put opto-couplers in place to protect both ends. Fibre/ethernet transceivers at both ends with a short run of fibre will protect both ends. Lightening strikes are only one problem, look to see what happens when one building attempts to ground itself through the copper cable to the other side. I would also question the legality of connecting both building with what I assume is mains electricity. B -- http://www.mailtrap.org.uk/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GPS-enabled cell phone/PDA
Hi There, this is very much dependent from your provider, your PDA/cell phone and the network. For GSM networks in Europe e.g. the providers have different types of information available through the CB channels of their base stations. This data can always be read and stored in your SMARTPHONE/PDA and when that has GPS data, then this data as well ... One nice examples are celltrack or gsmmon9210 for SYMBIAN based phones. What you do on the phone with the data is your business ;-) .. There are web-based databases available which show the exact location of the next station you're connected to. If you have GPS locally, than you have not to rely on thie cell data. Cell data inside cities can give your location as exact as to 100m, in rural areas it can be up to 5 km I suppose. Of course you can send the received data via SMS to other systems or with GPRS or WLAN access more or less online to Internet based services. With TDMA or IDEN phone systems which are used outside of Europe I have no experiences at all, sorry ... -- Jürgen I would like to capture the lat/lon coordinates from a GPS-enabled cell phone or PDA. Is this possible? Must I subscribe to this information from the cellphone network provider, or can I capture it without charge? What devices will broadcast the coordinates? Is there a device that will broadcast its position inband that can be captured by Asterisk? Can an SMS message include coordinates? The subject will willingly carry the device and will be aware that his location is being monitored, so privacy rights are not an issue. The subject will make periodic calls to the Asterisk server in order to record his movements. Does anyone have experience in this area? Its my understanding the cell phone coordinates are sent to the cell phone provider and their equipment reads (and holds) that data. Its not part of any data available to you in any form unless you talk to the cell provider and convience them you have a valid need. Highly unlikely in the US anyway. Even if you could convience them to provide it, they would likely demaand some sort of out-of-band data transmission facility. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What business IP phone to use
I have seen some very expensive switches fail. Nice thing about lower cost devices is that you can afford to have spares. If you stick to a standard way of labeling and connecting wires you can use good open source monitoring software to detect switch failure. If you allow people to randomly connect to a bank of switches it is not so easy to quickly find and remedy such problems. The more expensive switches are good if you are going to take advantage of the features they offer. I have recently seen situations like employees installing things like camera and itunes software that caused local network problems. Managed switches allowed immediate remote disconnection of the workstations. At this customer site the fancy switches are used for all workstations and some 3rd party servers(security video system is a good example). However, the customer-owned servers I installed are plugged into a $40 switch. Those servers are properly managed so there is no need for the features found in the more expensive switches. David Ankers wrote: Aha, micro seconds in networking terms is normally written usecs or us (actually it's the greek letter mu as in ulaw) rather than ms which are milliseconds seconds - what had me puzzled was that it was stated that this could harm the voice path! The difference can also cause unnecessary delays and therefor echo in the path. For example, procurve switches typically have 13ms switching time, the high-end netgears about 21ms. As soon as you stack a couple of switches you are talking 26ms vs 42ms extra delay in the path! There is then only 8 usecs between the two switches, how on earth would this make any difference to the voice path at all? Let alone induce any echo... Obviously the originally poster didn't understand the difference. And based on this, he's probably advising people not to use Netgear switches for voice, oh dear. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Watkins, Bradley Sent: Friday, 24 February 2006 10:08 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] What business IP phone to use It must be microseconds that is being quoted, as even the 2626 that you mention lists a less than 13.3 microsecond latency. - Brad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Ankers Sent: Thursday, February 23, 2006 6:54 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] What business IP phone to use Are you sure those switch figures are right? 16ms delay in the switch path sounds a bit long. Cisco's mid-range switches like the 2950 have switching times measured in micro seconds. Then again a 2626 procurve is only around $700. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Conrad Wood Sent: Friday, 24 February 2006 7:50 AM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] What business IP phone to use Simple formula: 1. Total Revenue 2. % of revenue derived from phone usage 3. =Cost of downtime by using SoHo or consumer gear. It's not a question of if a SoHo or low cost device will screw up, it is a question of when. This is 23 years of experience talking. Where I work, the value of #3 above is $16 Cdn a *second*. We are below 500 employees, so we fall into the SMB segment. Sometimes I'm appalled by statements that a $700 switch or a $400 phone isn't worth it. Huh?? Maybe in Absolutely right! for something as critical as switches cabling I always recommend to spend real money. Don't ever try to save money any equipment that is required to operate the business. (Had very good experience with HP procurves over the last 10 years or so). There is no point buying netgear or other low-cost switches for a business ever. The cost saving of being able to pin-point a cabling/NIC/bandwidth problem down to the port on the switch easily and quickly is wonderful. Combined with SNMP and all the other goodies good switches come with, our clients save a lot of money by paying me less for my time ( d'oh ;-) ). The difference can also cause unnecessary delays and therefor echo in the path. For example, procurve switches typically have 13ms switching time, the high-end netgears about 21ms. As soon as you stack a couple of switches you are talking 26ms vs 42ms extra delay in the path! I see no reason however to spend $400 on a single phone though, because if a single phone breaks, it's not going to bring your business to a standstill, is it? (I guess unless you only have one in the first place ;-) ) conrad ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Analyzer for Milliwatt
Andrew Kohlsmith wrote: On Friday 24 February 2006 07:56, Paul wrote: Maybe the first approach should be to setup a test extension for recording the tone. The idea is to get best resolution possible in real time. Then process it as much as needed to get the info you want. Such an approach would give you more flexibility. For example, you could automatically place periodic test calls to various servers and have the recordings then forwarded to one server for analysis. That would minimize the impact on production asterisk servers. What is being discussed here is basically what I was planning on doing for an automatic VOIP quality check. Using miliwatt and analyzing it for pop/jitter/etc as well as sending other known waveforms and comparing what was received to what was expected and coming up with some quality number which would be fed back to the dialplan to adjust the least-cost routing paths. Essentially come up with a least cost but still good quality routing. :-) I've done absolutely nothing other than a little research and a lot of thinking about how to do it though. I did some research on digital click/pop removal for records as a way to detect poor quality, and then also some monkeying around with coppice's excellent DSP routines in spandsp. I guess the best information would be obtained by recording in the codec format. That means being sure to prevent transcoding. I'm not sure if that can be done with simple dialplan programming. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] S100U and TigerJet
Hi all, this is another post about this problem. I installed from scratch a new Suse Linux 10.0, with latest stable asterisk. Moreover I add the lines to /etc/udev/rules.d/50-udev.rules, in order to let the driver create the /dev/zap... When I plug into usb port my TigerJet adapter, I see on /var/log/messages Feb 24 14:55:02 srvlnx05 kernel: usb 1-2: new full speed USB device using uhci_hcd and address 2 Feb 24 14:55:03 srvlnx05 kernel: usbcore: registered new driver snd-usb-audio Feb 24 14:55:03 srvlnx05 kernel: zaptel: module not supported by Novell, setting U taint flag. Feb 24 14:55:03 srvlnx05 kernel: Zapata Telephony Interface Registered on major 196 Feb 24 14:55:03 srvlnx05 kernel: wcusb: module not supported by Novell, setting U taint flag. Feb 24 14:55:03 srvlnx05 kernel: usbcore: registered new driver wcusb Feb 24 14:55:03 srvlnx05 kernel: Wildcard USB FXS Interface driver registered while lsusb shows Bus 001 Device 002: ID 06e6:831c Tiger Jet Network, Inc. Bus 001 Device 001: ID : under /dev, I see borning /zap and children srvlnx05:/etc # dir /dev/zap/ drwxr-xr-x 2 root root 120 Feb 24 14:55 . drwxr-xr-x 14 root root15720 Feb 24 14:55 .. crw-rw 1 asterisk asterisk 196, 254 Feb 24 14:55 channel crw-rw 1 asterisk asterisk 196, 0 Feb 24 14:55 ctl crw-rw 1 asterisk asterisk 196, 255 Feb 24 14:55 pseudo crw-rw 1 asterisk asterisk 196, 253 Feb 24 14:55 timer but NO channel 01 al all. I would like to know if anybody 1) ever succeded in having this configuration up and running. 2) ever succeded in having this configuration up and running with a *TRUE* S100U adapter from Digium. 3) If 2 is true *WHERE* it could be possible to buy this true adapter: on digium shop I was not able to find it. My opinion is that it could be an issue related to the operating system: I think I should do something similar to what I did on /etc/udev/rules.d/50-udev.rules in order to allow the creation of usb-related devices under /dev/zap. Unfortunately I don't know anything about Linux kernel enumeration process. Also, does exist any debugging tool for wcusb ? Wcusb is up and running, is the only in the system ( I removed the wcusb.ko natively present under the /extra directory) lsmod | grep wcu shows: srvlnx05:~ # lsmod | grep wcu wcusb 19104 0 zaptel187268 1 wcusb usbcore 112512 5 wcusb,snd_usb_audio,snd_usb_lib,uhci_hcd thank's all for attention. Andrea Chi ricevesse questa mail per errore e' gentilmente pregato di cancellarla. Visitate il sito http://www.frameweb.it ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] What business IP phone to use
On Sat, 2006-02-25 at 00:21 +1100, David Ankers wrote: Aha, micro seconds in networking terms is normally written usecs or us (actually it's the greek letter mu as in ulaw) rather than ms which are milliseconds seconds - what had me puzzled was that it was stated that this could harm the voice path! The difference can also cause unnecessary delays and therefor echo in the path. For example, procurve switches typically have 13ms switching time, the high-end netgears about 21ms. As soon as you stack a couple of switches you are talking 26ms vs 42ms extra delay in the path! There is then only 8 usecs between the two switches, how on earth would this make any difference to the voice path at all? Let alone induce any echo... Obviously the originally poster didn't understand the difference. And based on this, he's probably advising people not to use Netgear switches for voice, oh dear. Agree , previous statement was incorrect and I should probably not post late at night ;-) A few microseconds delay in the path obviously doesn't cause extra echo. Thank you for pointing that out. == Conrad ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: VoIP over bonded link
On Thursday 23 February 2006 13:57, Bob Goddard wrote: It's stupid. Don't ever connect 2 different building with copper. Just wait until you get some kind of lightening hit or electrical fault, but make sure you are no where near it. Use fibre. That's a great rule of thumb, but the reality isn't quite so black and white. A direct lightning strike is not going to draw *any* significant current through the ethernet cable, as the moment you try to pull significant current, those cables will either open up or vaporize due to IR losses in such small gage wire. You'll have far more current draw through the (I'm assuming) metal conduit, which is already grounded. Yes, you may introduce grounding loops and these will cause other (sometimes significant) issues but they have all been solved before. The best solution is to simply take a pair of media converters with a fiber patch cable between them, space them out adequately and hope for the best. You're already going to have a conduction path through the power supplies of the media converters but with an isolation transformer and appropriate surge arrestors it's about as best as you are going to be able to do. Electrical faults are *easily* dealt with with appropriate fusing, surge arrestors, isolation and plain old common sense. I work in the power electronics industry; we regularly deal with lightning strikes (both direct and close call style) and while there is very little to protect you from a direct strike (we use station-class arrestors) there is a LOT you can do to minimize grounding or loop problems when wiring between buildings. Sometimes fiber just doesn't cut it, so no, it's not just stupid. -A. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] fax receive using TDM400P
Hi! I am using tdm400 cards for receiving faxes. It worked quite out of the box. I installed spandsp for the rxfax application only. I use it as phone/fax switch: All incoming calls are answered automatically to listen whether its a fax or not. If it is a fax, the call is forwarded to the buil-in fax extension, otherwise the analog phones (all on tdm400) rings. It works without problems. Its for a small company (about a few faxes per hour) Tom Am Freitag, 24. Februar 2006 07:10 schrieb Anton Krall: Guys. Ive been testing how to receive faxes using TDM400P cards and so far, after playing with gains, echocancell and echotraining on zapata.conf.. Ive ha dno luck, faxes come in as garbage or broken or with blank lines. Anybody has successfully done this? Any tips.. Also I have some ideas: 1. Is it really possible to get fxes on a fax machine using ATAs like the sipura 2002? Even using ulaw and pass-thru, is it possible? 2. Since the faxes is coming from PSTN thru the card, I guess asterisk will always stay in the middle right? No way around this. 3. Im also trying to receive faxes usign a TE110P card with spandsp, unicall and E1 R2MFC, no luck also, some stuff, garbage and broken faxes. Anybody done this sucessfuly? Hope anybody can share their thoughts and insight on this. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thomas Artner ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] S100U and TigerJet
udev drove me absolutely bat-shit in this regard; udev is a horror in many respects. Here's how I solved the problem, reliably: I run this script at boot-time: #!/bin/bash mkdir -p /dev/zap rm -f /dev/zap/ctl rm -f /dev/zap/channel rm -f /dev/zap/pseudo rm -f /dev/zap/timer rm -f /dev/zap/253 rm -f /dev/zap/252 rm -f /dev/zap/251 rm -f /dev/zap/250 mknod /dev/zap/ctl c 196 0 mknod /dev/zap/timer c 196 253 mknod /dev/zap/channel c 196 254 mknod /dev/zap/pseudo c 196 255 N=1; \ while [ $N -lt 250 ]; do \ rm -f /dev/zap/$N; \ mknod /dev/zap/$N c 196 $N; \ N=$[$N+1]; \ done Have had zero problems with this. On Fri, 24 Feb 2006, [EMAIL PROTECTED] wrote: Hi all, this is another post about this problem. I installed from scratch a new Suse Linux 10.0, with latest stable asterisk. Moreover I add the lines to /etc/udev/rules.d/50-udev.rules, in order to let the driver create the /dev/zap... When I plug into usb port my TigerJet adapter, I see on /var/log/messages Feb 24 14:55:02 srvlnx05 kernel: usb 1-2: new full speed USB device using uhci_hcd and address 2 Feb 24 14:55:03 srvlnx05 kernel: usbcore: registered new driver snd-usb-audio Feb 24 14:55:03 srvlnx05 kernel: zaptel: module not supported by Novell, setting U taint flag. Feb 24 14:55:03 srvlnx05 kernel: Zapata Telephony Interface Registered on major 196 Feb 24 14:55:03 srvlnx05 kernel: wcusb: module not supported by Novell, setting U taint flag. Feb 24 14:55:03 srvlnx05 kernel: usbcore: registered new driver wcusb Feb 24 14:55:03 srvlnx05 kernel: Wildcard USB FXS Interface driver registered while lsusb shows Bus 001 Device 002: ID 06e6:831c Tiger Jet Network, Inc. Bus 001 Device 001: ID : under /dev, I see borning /zap and children srvlnx05:/etc # dir /dev/zap/ drwxr-xr-x 2 root root 120 Feb 24 14:55 . drwxr-xr-x 14 root root15720 Feb 24 14:55 .. crw-rw 1 asterisk asterisk 196, 254 Feb 24 14:55 channel crw-rw 1 asterisk asterisk 196, 0 Feb 24 14:55 ctl crw-rw 1 asterisk asterisk 196, 255 Feb 24 14:55 pseudo crw-rw 1 asterisk asterisk 196, 253 Feb 24 14:55 timer but NO channel 01 al all. I would like to know if anybody 1) ever succeded in having this configuration up and running. 2) ever succeded in having this configuration up and running with a *TRUE* S100U adapter from Digium. 3) If 2 is true *WHERE* it could be possible to buy this true adapter: on digium shop I was not able to find it. My opinion is that it could be an issue related to the operating system: I think I should do something similar to what I did on /etc/udev/rules.d/50-udev.rules in order to allow the creation of usb-related devices under /dev/zap. Unfortunately I don't know anything about Linux kernel enumeration process. Also, does exist any debugging tool for wcusb ? Wcusb is up and running, is the only in the system ( I removed the wcusb.ko natively present under the /extra directory) lsmod | grep wcu shows: srvlnx05:~ # lsmod | grep wcu wcusb 19104 0 zaptel187268 1 wcusb usbcore 112512 5 wcusb,snd_usb_audio,snd_usb_lib,uhci_hcd thank's all for attention. Andrea Chi ricevesse questa mail per errore e' gentilmente pregato di cancellarla. Visitate il sito http://www.frameweb.it ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Possible Bug in SIP Stack.
I currently use asterisk version 1.0.10 with AMP 1.0.010, our setup is APX 8000 - Interaction SIP Proxy 3.0.013 - asterisk server. When I use Asterisk version 10.0.10 everything works perfectly, however when I use 1.2.4 I lose the ability to receive calls from the PSTN. All I get is the following error in my SIP Proxies error logs: SIPSession::proxyResponseImmediately(): Failed to retrieve next Via, don't know where to send responseSIP/2.0 180 Ringing From: MODESITT,CHRIS sip:[EMAIL PROTECTED]:5060;user=phone;tag=4fdc9d0e-1e600f94-ed7e623f To: sip:[EMAIL PROTECTED]:5060;user=phone;tag=as4fc8aa8a Call-ID: [EMAIL PROTECTED] CSeq: 5466974 INVITE User-Agent: Asterisk PBX I still can make outbound calls with no-problems, any ideas? Thanks Chris ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with T1 installation
Nitin Joshi wrote: Hi All, I have installed a Digium TE110P card on an Asterisk 1.2.1 system. Its connected directly to the PSTN. But I am unable to make outbound calls on the zap channels. The light on the card is green. Asterisk CLI shows all 24 channels when I give the command 'zap show channels'. I also noticed that Asterisk CLI shows an incoming call every few seconds on the 24th channel. This must be some kind of a timing signal. This is he first time I am configuring a T1 so I must have done something wrong I guess. T1s require a D (Data) channel, unless connecting to a channel bank, It should be 23 voice 1 data. Also, I would strongly suggest moving to 1.2.4 Doug ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Explain Yellow Alarm in a Legacy Integration
On 2/23/06, Geoff Manning [EMAIL PROTECTED] wrote: How would you categorize a Yellow Alarm sensed by the Asterisk side in a Legacy PBX integration?We have a Mitel SX200 connected to an Asterisk(1.2.4) with a TE110P.Twice today (first time in over a month) we received a Yellow Alarm on the TE110P. I have been able to clear it easily by restarting zaptel. Thanks in advance! So we had another Yellow Alarm last night and I have retrieved the logs from the Mitel. It had a Red Alarm.Here seems to be the order of events:Mitel PBX:³2006-FEB-24 02:44:54 T1/BRI card at 02 06 00 00 ³ ³ has exceeded the service loss frame threshold ³³2006-FEB-24 02:44:54 Tot alarm went from No Alarm to MAJOR ³³ Alarm level change due to Bay 02 trunks ³ ³2006-FEB-24 02:44:54 T1/BRI card at 02 06 00 00 ³³ removed from service transmitting yellow alarm ³Asterisk:Feb 24 02:45:43 WARNING[24210] chan_zap.c: Detected alarm on channel 1: Yellow Alarm Mitel PBX (This is when we manually reset the card on the Asterisk to clear the alarm):³2006-FEB-24 05:25:45 T1/BRI card at 02 06 00 00 ³³ is in red alarm condition due to loss of sync ³ ³2006-FEB-24 05:26:08 T1/BRI card at 02 06 00 00 ³³ alarm condition is now cleared ³³2006-FEB-24 05:26:08 Tot alarm went from MAJOR to No Alarm ³ ³ Alarm level change due to Bay 02 trunks Asterisk:Feb 24 05:26:54 NOTICE[24210] chan_zap.c: Alarm cleared on channel 1So it seems the Mitel is reaching a loss threshold and setting yellow alarm. Asterisk is in turn detecting the yellow alarm. I guess it's a problem with the Mitel then. We've had problems with it in the past but they cleared up and we hadn't had an issue in months. Nothing has changed at either end but we've been hit with issues for the last 3 days. Here is what I have found about the alarms:Red Alarm This is a local equipment alarm. It indicates that the incoming signal has been corrupted for a number of seconds. The red alarm shows up visually on the equipment that detects the failure. This equipment will then begin sending a yellow alarm as its outbound signal. Yellow Alarm The yellow alarm alerts the network that a failure has been detected. The yellow alarm pattern has a number of different definitions. The most common D4 definition is to set 1 bit of every channel to a ZERO. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: [asterisk-dev] Possible Bug in SIP Stack.
Chris Modesitt wrote: I currently use asterisk version 1.0.10 with AMP 1.0.010, our setup is APX 8000 - Interaction SIP Proxy 3.0.013 - asterisk server. When I use Asterisk version 10.0.10 everything works perfectly, however when I use 1.2.4 I lose the ability to receive calls from the PSTN. All I get is the following error in my SIP Proxies error logs: SIPSession::proxyResponseImmediately(): Failed to retrieve next Via, don't know where to send responseSIP/2.0 180 Ringing From: MODESITT,CHRIS sip:[EMAIL PROTECTED]:5060;user=phone;tag=4fdc9d0e-1e600f94-ed7e623f To: sip:[EMAIL PROTECTED]:5060;user=phone;tag=as4fc8aa8a Call-ID: [EMAIL PROTECTED] CSeq: 5466974 INVITE User-Agent: Asterisk PBX I still can make outbound calls with no-problems, any ideas? Can you get SIP debug logs from a call setup with Asterisk 1.0.10 and 1.2.4 so we can compare them and see what happened? Thanks /Olle ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GPS-enabled cell phone/PDA
On Fri, Feb 24, 2006 at 07:17:52AM -0600, Rich Adamson wrote: Its my understanding the cell phone coordinates are sent to the cell phone provider and their equipment reads (and holds) that data. Its not part of any data available to you in any form unless you talk to the cell provider and convience them you have a valid need. Highly unlikely in the US anyway. Even if you could convience them to provide it, they would likely demaand some sort of out-of-band data transmission facility. GSM networks have the Cell ID available to the phone, however that's not much use without the location of the cellsite. There are now location based services, whereby you can query the network and they'll give out an approximate location (most cells are sectored [6 sectors per cell) which gives a direction, the cell also knows what power the phone is transmitting with, and the power it's received so can make a good approximation of where the phone is (within 60 degrees angle). However it's likely a phone will be picked up by several cells, so the network can triangulate and make a better aproximation. Making the information available to end-users is problematic due to privacy issues, unless the user explicitly agrees to give the info away. With GPS units, the info is stored in the phone and can send it out using SMS or other means. Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] What business IP phone to use
On Wed, 22 Feb 2006 18:02:27 -0800, mustardman29 wrote: Just the person I have been looking for. If you don't mind, would it be possible to get your opinion on feature for feature comparisons between the 501 and 480i CT(not including cordless phone). Things like programmable buttons, display, dialing button quality, and most importantly, handset and speakerphone quality. Any info would be greatly appreciated. I used the IP600 for about a year on my desk, and several IP500s elsewhere around the place. It's a home office but I work from home full time so it's a real working office environment. I found that the physical quality of the Polycom phones was absolutely top notch. They're a joy to use. Completely professional and very reliable. But they're not perfect. They're a little harder to provision. They're very configurable but that also adds to the complexity. I had mine TFTP loading firmware and a common speed dial directory from an XML file on my Astlinux server. The phones take a fair amount of time to boot and force a reboot when you change many of their settings. You can spend an afternoon repeatedly rebooting the phone as you manually work out its initial configuration. Of course Polycom doesn't support Asterisk, but others seem to fill this void well enough. The IP600 and IP500 are very similar but the differences are considerable. The IP600 supports 6 line buttons and has a much better LCD. Higher resolution, but still not backlit. Once you've used the 600 it'll be hard to go back to the 500 just because the display is not as nice. The IP500 provides only 3 line buttons. Both phones support multiple registrations. The Aastra 480 is the only thing that I've seen that comes close to the Polycom's. Physically it's just about as solid. Not quite as hefty in the hand, but very nice. The LCD display is backlit. This is a major advantage if you ever work in dim lighting. All other manufacturers...LISTEN UP...this is a really big deal! I can't believe how long its taken for someone to realise this fact. Aastra configuration was a LOT easier both manually on the phone and remotely. The on-phone menus are very easy to navigate and I almost didn't bother setting up the central provisioning. With only a few phones I could get by without it. Firmware and configs can be loaded via tftp, ftp or http. The on-phone directory and call logs are comparable on all three the I have used. Actually, I prefer the way SNOM phones handle this as they require fewer button presses. The Aastra phone makes it especially easy to delete an entire call log with only a couple of button presses. The 480 supports up to 9 lines with any 4 active at on time, or so I'm told. I have mine registered for four lines so that incomming PSTN, FWD, Gizmo and Skype calls each ring a different line. The latest firmware supposedly support BLF indications but I've not used this. It's really easy to assign speed dials to the six programmable keys on the LCD. In fact, almost all of the buttons can be reassigned to new functions. Also you can write XML applications that put the LCD to work as an interactive menu. Mostly I live and die by speakerphone quality. I think that the Polycom's have a little edge on the Aastra phone, but not by much. If I need to rework my entire system I'll probably migrate to all Aastra phones. Audio quality using the handset is excellent on all of them. Even on the cordless handset with the 480i CT. They all support POE...which I use to keep the phone system up during power failures. I had to buy the injectors separately for the Aastra IP600 phones. The IP500s came with injector cables. The big dissappointment in my SIP phone testing was the Zultys 4x5. It just feels cheap and many functions are too counterintuitive. I really like the idea of the local FXO but they were never able to tell me how to get the FXO port forwarded to the PBX for VM. Zultys provides no end user support except through dealers and the dealers I dealt with didn't know much about the specifics of the Zultys firmware. Also, I'm curious about the newest SNOM phones. Some time ago I used a SNOM 200 and like the way the web based I/F was integrated into the use of the phone beyond simply configuration. You could access the speed dials and place a call from the web I/F. You could also dial the phone from a link or shortcut to a url pointed at the phone. That's a fair substitute for desktop TAPI. If they've taken this any further it could be very good. I've not tried any of the lesser phones like Grandstream or Linksys. Life's too short to use a cheap phoneat least if your budget permits better. Michael Graves -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 fwd 54245 ___ --Bandwidth and Colocation
[Asterisk-Users] Re: [asterisk-dev] Possible Bug in SIP Stack.
Chris Modesitt wrote: I currently use asterisk version 1.0.10 with AMP 1.0.010, our setup is APX 8000 - Interaction SIP Proxy 3.0.013 - asterisk server. When I use Asterisk version 10.0.10 everything works perfectly, however when I use 1.2.4 I lose the ability to receive calls from the PSTN. All I get is the following error in my SIP Proxies error logs: SIPSession::proxyResponseImmediately(): Failed to retrieve next Via, don't know where to send responseSIP/2.0 180 Ringing From: MODESITT,CHRIS sip:[EMAIL PROTECTED]:5060;user=phone;tag=4fdc9d0e-1e600f94-ed7e623f To: sip:[EMAIL PROTECTED]:5060;user=phone;tag=as4fc8aa8a Call-ID: [EMAIL PROTECTED] CSeq: 5466974 INVITE User-Agent: Asterisk PBX I still can make outbound calls with no-problems, any ideas? Can you get SIP debug logs from a call setup with Asterisk 1.0.10 and 1.2.4 so we can compare them and see what happened? Thanks /Olle ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] fax receive using TDM400P
I wrestled with this for a long time, as have many others and it just doesn't work with spandsp and asterisk alone. Use iaxmodem and hylafax in conjunction with asterisk... it works like a champ. I have a single POTS line coming in so I get voice fax with a single number using fax detect. http://iaxmodem.sourceforge.net/ -Original Message- From: Rich Adamson [mailto:[EMAIL PROTECTED] Sent: Friday, February 24, 2006 7:28 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] fax receive using TDM400P Ive been testing how to receive faxes using TDM400P cards and so far, after playing with gains, echocancell and echotraining on zapata.conf.. Ive ha dno luck, faxes come in as garbage or broken or with blank lines. Anybody has successfully done this? Any tips.. Also I have some ideas: 1. Is it really possible to get fxes on a fax machine using ATAs like the sipura 2002? Even using ulaw and pass-thru, is it possible? 2. Since the faxes is coming from PSTN thru the card, I guess asterisk will always stay in the middle right? No way around this. 3. Im also trying to receive faxes usign a TE110P card with spandsp, unicall and E1 R2MFC, no luck also, some stuff, garbage and broken faxes. Anybody done this sucessfuly? Hope anybody can share their thoughts and insight on this. Using the TDM400 card for any form of fax'ing (or modem use) is well known to be unreliable and, in most cases, totally unusable. The issue has been discussed many times over the last two years or so. There are no known workarounds. Its my understanding that lots of folks have spandsp working via T1 and/or PRI interfaces. The issues associated with the TDM400 card do not apply to the T1 cards. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GPS-enabled cell phone/PDA
In the US, Sprint's CDMA network will do the fancy GPS+AFLT business, but like someone else mentioned, it only sends the location data back to Sprint's network. There is an API that you can use to access this data for your handsets, but you have to pay some amount of money for each location fix. Sprint's iDEN phones (formerly Nextel) contain GPS units that can be accessed from the phone's serial port, and I am pretty sure that the GPS data can be accessed from a J2ME applet running in the phone. Such an applet could then make an appropriate HTTP request to a web/app server you run, in order to upload the data. However, the GPS data received using this method is obtained using _only_ GPS, with no AFLT or other form of assistance from the cellular network. The significance of that, of course, is that you will not be able to get a GPS fix in locations where a regular GPS receiver can't get a fix, such as indoors in most cases. -Rusty On 2/23/06, Michael Welter [EMAIL PROTECTED] wrote: I would like to capture the lat/lon coordinates from a GPS-enabled cell phone or PDA. Is this possible? Must I subscribe to this information from the cellphone network provider, or can I capture it without charge? What devices will broadcast the coordinates? Is there a device that will broadcast its position inband that can be captured by Asterisk? Can an SMS message include coordinates? The subject will willingly carry the device and will be aware that his location is being monitored, so privacy rights are not an issue. The subject will make periodic calls to the Asterisk server in order to record his movements. Does anyone have experience in this area? Thanks, Mike -- Michael Welter Telecom Matters Corp. Denver, Colorado US +1.303.414.4980 [EMAIL PROTECTED] www.TelecomMatters.net ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Beer meeting at Fosdem
Hi Olle, Will u be there for the speech of Jan Janak? If yes, you will find a guy, 1m83, with a bear and a red suit, it's me. You also can call me on my mobile to fix the voip beer (0032495283361). We will try to have Jan and other guys Olivier ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] S100U and TigerJet
no chance, also with your scipt ztcfg -vvv Zaptel Configuration == Channel map: Channel 01: FXO Kewlstart (Default) (Slaves: 01) 1 channels configured. ZT_CHANCONFIG failed on channel 1: No such device or address (6) Jerry Glomph Black [EMAIL PROTECTED] To lomph.comAsterisk Users Mailing List - Non-Commercial Discussion 24/02/2006 15.29 asterisk-users@lists.digium.com cc [EMAIL PROTECTED] Subject Re: [Asterisk-Users] S100U and TigerJet udev drove me absolutely bat-shit in this regard; udev is a horror in many respects. Here's how I solved the problem, reliably: I run this script at boot-time: #!/bin/bash mkdir -p /dev/zap rm -f /dev/zap/ctl rm -f /dev/zap/channel rm -f /dev/zap/pseudo rm -f /dev/zap/timer rm -f /dev/zap/253 rm -f /dev/zap/252 rm -f /dev/zap/251 rm -f /dev/zap/250 mknod /dev/zap/ctl c 196 0 mknod /dev/zap/timer c 196 253 mknod /dev/zap/channel c 196 254 mknod /dev/zap/pseudo c 196 255 N=1; \ while [ $N -lt 250 ]; do \ rm -f /dev/zap/$N; \ mknod /dev/zap/$N c 196 $N; \ N=$[$N+1]; \ done Have had zero problems with this. On Fri, 24 Feb 2006, [EMAIL PROTECTED] wrote: Hi all, this is another post about this problem. I installed from scratch a new Suse Linux 10.0, with latest stable asterisk. Moreover I add the lines to /etc/udev/rules.d/50-udev.rules, in order to let the driver create the /dev/zap... When I plug into usb port my TigerJet adapter, I see on /var/log/messages Feb 24 14:55:02 srvlnx05 kernel: usb 1-2: new full speed USB device using uhci_hcd and address 2 Feb 24 14:55:03 srvlnx05 kernel: usbcore: registered new driver snd-usb-audio Feb 24 14:55:03 srvlnx05 kernel: zaptel: module not supported by Novell, setting U taint flag. Feb 24 14:55:03 srvlnx05 kernel: Zapata Telephony Interface Registered on major 196 Feb 24 14:55:03 srvlnx05 kernel: wcusb: module not supported by Novell, setting U taint flag. Feb 24 14:55:03 srvlnx05 kernel: usbcore: registered new driver wcusb Feb 24 14:55:03 srvlnx05 kernel: Wildcard USB FXS Interface driver registered while lsusb shows Bus 001 Device 002: ID 06e6:831c Tiger Jet Network, Inc. Bus 001 Device 001: ID : under /dev, I see borning /zap and children srvlnx05:/etc # dir /dev/zap/ drwxr-xr-x 2 root root 120 Feb 24 14:55 . drwxr-xr-x 14 root root15720 Feb 24 14:55 .. crw-rw 1 asterisk asterisk 196, 254 Feb 24 14:55 channel crw-rw 1 asterisk asterisk 196, 0 Feb 24 14:55 ctl crw-rw 1 asterisk asterisk 196, 255 Feb 24 14:55 pseudo crw-rw 1 asterisk asterisk 196, 253 Feb 24 14:55 timer but NO channel 01 al all. I would like to know if anybody 1) ever succeded in having this configuration up and running. 2) ever succeded in having this configuration up and running with a *TRUE* S100U adapter from Digium. 3) If 2 is true *WHERE* it could be possible to buy this true adapter: on digium shop I was not able to find it. My opinion is that it could be an issue related to the operating system: I think I should do something similar to what I did on /etc/udev/rules.d/50-udev.rules in order to allow the creation of usb-related devices under /dev/zap. Unfortunately I don't know anything about Linux kernel enumeration process. Also, does exist any debugging tool for wcusb ? Wcusb is up and running, is the only in the system ( I removed the wcusb.ko natively present under the /extra directory) lsmod | grep wcu shows: srvlnx05:~ # lsmod | grep wcu wcusb 19104 0 zaptel187268 1 wcusb usbcore 112512 5 wcusb,snd_usb_audio,snd_usb_lib,uhci_hcd
RE: [Asterisk-Users] spandsp debug or logging
Done.. They don't show much but they do show some problems with lost lines or something Thx Bartosz |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Bartosz Piec |Sent: Friday, February 24, 2006 2:54 AM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] spandsp debug or logging | |Anton Krall wrote: | Maybe this is a stupid question but how to you enable debubg or | logging on spandsp? I see you can do that for RXFAX but when people | tell you to enable debug on spandsp, do they mean this with rxfax or | how do you do it with spandsp? | |You can do it writing: | |exten = s,1,rxfax(/fax/file/path|debug) | |or the same with txfax. The logs are then written to (default) |/var/log/asterisk/full | |-- |Best regards, |Bartosz Piec |___ |--Bandwidth and Colocation provided by Easynews.com -- | |Asterisk-Users mailing list |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | | ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: What business IP phone to use
maybe you didn't want suggestions, but too bad :). My favorite up until recently was the polycom 501 and I found it was good quality and clear calls and priced well. but the production of te phone is slowing down so I bought a few linksysspa941. and iVll tell you I have a new favorite phone. its slick, provisioning is a breeeze and the call quality with built in qos is fantastic. I wasn't a big fan of grandstream products they seem to be cheaply made and i've had a few fail. but they do work. talking about my biased opinion I don't have onee, i'm a hobby programmer who works for a company that resells voip services and we use polycom and linksys. I just provide support for all phones so I kow how things work and don't work. I hope this helps. thanks andrew On 2/21/06, mustardman29 [EMAIL PROTECTED] wrote: I have been struggling with this issue for about a year now. There were just too many IP phones to choose from at all sorts of price points and not enough information about any of them. Now I am looking at the situation again and if anything it has gotten worse. There are even more phones and all sorts of opinions. For every person that says phone x is great there is someone else complaining about it. I ended up buying a Grandstream GXP2000 and an Aastra 9133i to test so I pretty much know what those two phones are about. Lot's of people talking about Polycom phones but they still seem to have their problems and since they don't officially support Asterisk I have my concerns. I really don't want to have to keep buying phones to find out for myself as it get's expensive real fast. Is there any unbiased comparison of various phones and features anywhere. If someone wrote a book I'd buy it but it would probably be obsolete before it was published with the rate of new IP phone introductions and firmware revisons. I hear some people praising the GXP2000 phones and I gotta wonder what they are smokin (regardless of firmware revison) so I just don't know who to believe anymore. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Keep getting message in logs that pbx.c cannot find extension context 'default'
do you have a defaultcontext=something parameter in sip.conf [general] section?? If not, the default is... em default RegardsOn 2/23/06, Chuck Bunn [EMAIL PROTECTED] wrote: Hi,I am getting repeated messages in my logs with the following:*Feb 23 07:56:11 NOTICE[2470] pbx.c: Cannot find extension context 'default'Feb 23 07:56:11 DEBUG[2470] chan_sip.c: SIP message could not be handled, bad request: [EMAIL PROTECTED]Feb 23 07:56:12 NOTICE[2470] pbx.c: Cannot find extension context 'default' Feb 23 07:56:12 DEBUG[2470] chan_sip.c: SIP message could not behandled, bad request: [EMAIL PROTECTED]Feb 23 07:56:14 NOTICE[2470] pbx.c: Cannot find extension context 'default'Feb 23 07:56:14 DEBUG[2470] chan_sip.c: SIP message could not behandled, bad request: [EMAIL PROTECTED] *I do not have a default context used in my extensions.conf - I use othernames. Am I required to have a context named 'default'??Thanks___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] fax receive using TDM400P
Well, I have the same effect on my TDM as in the E1 using unicall... Faxes get here as garbage :( |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Rich Adamson |Sent: Friday, February 24, 2006 7:28 AM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] fax receive using TDM400P | | Ive been testing how to receive faxes using TDM400P cards |and so far, | after playing with gains, echocancell and echotraining on | zapata.conf.. Ive ha dno luck, faxes come in as garbage or |broken or with blank lines. | | Anybody has successfully done this? Any tips.. Also I have |some ideas: | | 1. Is it really possible to get fxes on a fax machine using |ATAs like | the sipura 2002? Even using ulaw and pass-thru, is it possible? | | 2. Since the faxes is coming from PSTN thru the card, I |guess asterisk | will always stay in the middle right? No way around this. | | 3. Im also trying to receive faxes usign a TE110P card with spandsp, | unicall and E1 R2MFC, no luck also, some stuff, garbage and broken | faxes. Anybody done this sucessfuly? | | Hope anybody can share their thoughts and insight on this. | |Using the TDM400 card for any form of fax'ing (or modem use) |is well known to be unreliable and, in most cases, totally |unusable. The issue has been discussed many times over the |last two years or so. There are no known workarounds. | |Its my understanding that lots of folks have spandsp working |via T1 and/or PRI interfaces. The issues associated with the |TDM400 card do not apply to the T1 cards. | | |___ |--Bandwidth and Colocation provided by Easynews.com -- | |Asterisk-Users mailing list |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | | ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] What business IP phone to use
Polycom does support Asterisk, Asterisk Business Edition. -Original Message- From: Michael Graves [mailto:[EMAIL PROTECTED] Sent: Thursday, February 23, 2006 6:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] What business IP phone to use On Wed, 22 Feb 2006 18:02:27 -0800, mustardman29 wrote: Just the person I have been looking for. If you don't mind, would it be possible to get your opinion on feature for feature comparisons between the 501 and 480i CT(not including cordless phone). Things like programmable buttons, display, dialing button quality, and most importantly, handset and speakerphone quality. Any info would be greatly appreciated. I used the IP600 for about a year on my desk, and several IP500s elsewhere around the place. It's a home office but I work from home full time so it's a real working office environment. I found that the physical quality of the Polycom phones was absolutely top notch. They're a joy to use. Completely professional and very reliable. But they're not perfect. They're a little harder to provision. They're very configurable but that also adds to the complexity. I had mine TFTP loading firmware and a common speed dial directory from an XML file on my Astlinux server. The phones take a fair amount of time to boot and force a reboot when you change many of their settings. You can spend an afternoon repeatedly rebooting the phone as you manually work out its initial configuration. Of course Polycom doesn't support Asterisk, but others seem to fill this void well enough. The IP600 and IP500 are very similar but the differences are considerable. The IP600 supports 6 line buttons and has a much better LCD. Higher resolution, but still not backlit. Once you've used the 600 it'll be hard to go back to the 500 just because the display is not as nice. The IP500 provides only 3 line buttons. Both phones support multiple registrations. The Aastra 480 is the only thing that I've seen that comes close to the Polycom's. Physically it's just about as solid. Not quite as hefty in the hand, but very nice. The LCD display is backlit. This is a major advantage if you ever work in dim lighting. All other manufacturers...LISTEN UP...this is a really big deal! I can't believe how long its taken for someone to realise this fact. Aastra configuration was a LOT easier both manually on the phone and remotely. The on-phone menus are very easy to navigate and I almost didn't bother setting up the central provisioning. With only a few phones I could get by without it. Firmware and configs can be loaded via tftp, ftp or http. The on-phone directory and call logs are comparable on all three the I have used. Actually, I prefer the way SNOM phones handle this as they require fewer button presses. The Aastra phone makes it especially easy to delete an entire call log with only a couple of button presses. The 480 supports up to 9 lines with any 4 active at on time, or so I'm told. I have mine registered for four lines so that incomming PSTN, FWD, Gizmo and Skype calls each ring a different line. The latest firmware supposedly support BLF indications but I've not used this. It's really easy to assign speed dials to the six programmable keys on the LCD. In fact, almost all of the buttons can be reassigned to new functions. Also you can write XML applications that put the LCD to work as an interactive menu. Mostly I live and die by speakerphone quality. I think that the Polycom's have a little edge on the Aastra phone, but not by much. If I need to rework my entire system I'll probably migrate to all Aastra phones. Audio quality using the handset is excellent on all of them. Even on the cordless handset with the 480i CT. They all support POE...which I use to keep the phone system up during power failures. I had to buy the injectors separately for the Aastra IP600 phones. The IP500s came with injector cables. The big dissappointment in my SIP phone testing was the Zultys 4x5. It just feels cheap and many functions are too counterintuitive. I really like the idea of the local FXO but they were never able to tell me how to get the FXO port forwarded to the PBX for VM. Zultys provides no end user support except through dealers and the dealers I dealt with didn't know much about the specifics of the Zultys firmware. Also, I'm curious about the newest SNOM phones. Some time ago I used a SNOM 200 and like the way the web based I/F was integrated into the use of the phone beyond simply configuration. You could access the speed dials and place a call from the web I/F. You could also dial the phone from a link or shortcut to a url pointed at the phone. That's a fair substitute for desktop TAPI. If they've taken this any further it could be very good. I've not tried any of the lesser phones like Grandstream or Linksys. Life's too short to use a cheap phoneat least if your budget permits better. Michael Graves -- Michael Graves
RE: [Asterisk-Users] fax receive using TDM400P
Any modification made to zapata as far as echo and gains? Should echocancel be on or off? |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Thomas Artner |Sent: Friday, February 24, 2006 8:25 AM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] fax receive using TDM400P | |Hi! | |I am using tdm400 cards for receiving faxes. It worked quite |out of the box. I installed spandsp for the rxfax application only. | |I use it as phone/fax switch: |All incoming calls are answered automatically to listen |whether its a fax or not. If it is a fax, the call is |forwarded to the buil-in fax extension, otherwise the analog |phones (all on tdm400) rings. | |It works without problems. Its for a small company (about a |few faxes per |hour) | | |Tom | | | | |Am Freitag, 24. Februar 2006 07:10 schrieb Anton Krall: | Guys. | | Ive been testing how to receive faxes using TDM400P cards |and so far, after | playing with gains, echocancell and echotraining on |zapata.conf.. Ive ha | dno luck, faxes come in as garbage or broken or with blank lines. | | Anybody has successfully done this? Any tips.. Also I have |some ideas: | | 1. Is it really possible to get fxes on a fax machine using |ATAs like the | sipura 2002? Even using ulaw and pass-thru, is it possible? | | 2. Since the faxes is coming from PSTN thru the card, I |guess asterisk will | always stay in the middle right? No way around this. | | 3. Im also trying to receive faxes usign a TE110P card with spandsp, | unicall and E1 R2MFC, no luck also, some stuff, garbage and |broken faxes. | Anybody done this sucessfuly? | | Hope anybody can share their thoughts and insight on this. | | ___ | --Bandwidth and Colocation provided by Easynews.com -- | | Asterisk-Users mailing list | To UNSUBSCRIBE or update options visit: |http://lists.digium.com/mailman/listinfo/asterisk-users | |-- |Thomas Artner |___ |--Bandwidth and Colocation provided by Easynews.com -- | |Asterisk-Users mailing list |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | | ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Is setting the variable _TRANSFER_CONTEXT required in features.conf?
you need to set a TRANSFER_CONTEXT, either for the transferer or transferee channel. I dont know why, but res_features give priority to the transferee TRANSFER_CONTEXT, if not found, then use the transferer TRANSFER_CONTEXT. That context is used to match the extension to dial. So you can set this var to any context you want. RegardsOn 2/23/06, Chuck Bunn [EMAIL PROTECTED] wrote: Hi,Is setting the variable _TRANSFER_CONTEXT required in features.conf forAsterisk 1.2.4? It appears from the Wiki that transfers across contextsare not possible when this is set. If it is not set can one trasfer across contexts??Thanks___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] GPS-enabled cell phone/PDA
In the UK this is common; several websites enable you to track a cell phone online: http://www.traceamobile.co.uk/ and another: http://www.followus.co.uk/ Works the same way that Steve stated... The police here in Australia have been using this since the late 90s. Interesting article: http://www.guardian.co.uk/g2/story/0,,1699080,00.html -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Kennedy Sent: Saturday, 25 February 2006 1:57 AM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] GPS-enabled cell phone/PDA On Fri, Feb 24, 2006 at 07:17:52AM -0600, Rich Adamson wrote: Its my understanding the cell phone coordinates are sent to the cell phone provider and their equipment reads (and holds) that data. Its not part of any data available to you in any form unless you talk to the cell provider and convience them you have a valid need. Highly unlikely in the US anyway. Even if you could convience them to provide it, they would likely demaand some sort of out-of-band data transmission facility. GSM networks have the Cell ID available to the phone, however that's not much use without the location of the cellsite. There are now location based services, whereby you can query the network and they'll give out an approximate location (most cells are sectored [6 sectors per cell) which gives a direction, the cell also knows what power the phone is transmitting with, and the power it's received so can make a good approximation of where the phone is (within 60 degrees angle). However it's likely a phone will be picked up by several cells, so the network can triangulate and make a better aproximation. Making the information available to end-users is problematic due to privacy issues, unless the user explicitly agrees to give the info away. With GPS units, the info is stored in the phone and can send it out using SMS or other means. Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] mpg123 alternative?
Thank you Lee, Dave, Rich, Joel and of course also Kevin. Between your various messages I finally understand what's happening and how it works, and have actually converted everything to alaw, ulaw, slin and gsm and am not actually using the mp3 side of things at all anymore. The difference is very noticeable in terms of MOH quality except when using g729 on the link between Asterisk and the phone - the sound quality seems worse there. I have two related questions though which I'm hoping someone can help with: We use alaw, ulaw, gsm and g729 between phones and asterisk. Sox can convert files to ulaw, alaw and gsm (not to mention slin) but what about g729? Is there such a thing as a format that won't need transcoding when using g729 links, or is this not something that is possible? And what is the signed linear (slin) format used for? Thanks, Faris. Lee Archer wrote: Check out the musiconhold.conf.sample in the asterisksource/configs folder. Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Faris Raouf Sent: 23 February 2006 18:36 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] mpg123 alternative? Ah! Now this is actually something I've not been able to get my head around: Note: As of Asterisk 1.2.0, Mpg123 is no longer used by Asterisk, which has its own MP3 player. Can anybody tell me where this built-in MP3 player is in 1.2.x/how do I use it ? I still seem to have the usual two mpg123 processes running with 1.2.4, with whatever music on hold is set in musiconhold.conf I'm sure it is very obvious, but I can't for the life of me figure out what I'm supposed to do to use the built-in MP3 player facilities. I just have the following in my musiconhold.conf: [default] mode=mp3 directory=/var/lib/asterisk/mohmp3 random=yes Faris. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Contact Center
I have talked with of a couple people(don't remember their names) who had this developed on a contract basis for the 1.0 Asterisk code tree, they did not want to release it to GPL because of how much it cost them and the fact that their code supposedly won't run on 1.2, but it is technically possible and has been talked about many times on the list. There was even a feature request for this over 2 years ago, it was dismissed as being too hard: http://bugs.digium.com/view.php?id=633 There was talk of this last month on the dev list: http://threebit.net/mail-archive/asterisk-dev/msg2.html Maybe it's time for somebody to organize a bounty for it: http://www.voip-info.org/wiki-Asterisk+bounty MATT--- On 2/24/06, Stephen Arulraj [EMAIL PROTECTED] wrote: Can the asterisk support a coaching function for the Supervisor to tap onto a call and coach the agent as she speaks to the customer without the customer hearing it.? Customer database management softward (or CRM) – is this being included? Best regards Stephen ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with T1 installation
On 2/24/06, Doug Lytle [EMAIL PROTECTED] wrote: T1s require a D (Data) channel, unless connecting to a channel bank, Itshould be 23 voice 1 data.Also, I would strongly suggest moving to 1.2.4 Not totally true. A PRI is 23b 1d. A DS1 (US) is a 24 channel circuit. Nitin- When you stop/start asterisk does it load all 24 channels? Any errors? How about zap show channel 1 in the CLI? -Brian ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960 (SIP) with Asterisk: how to get # towork during a call
Mike, Were you able to get this working? Even after with a entry in the dialplan.xml does not work for me. Thanks, Ken On 6/20/05, Michael J. Tubby B.Sc (Hons) G8TIC [EMAIL PROTECTED] wrote: Andrew,I presume you mean in the Cisco 7940/7960 SIP Phone Administrator's Guide?When you say mapped, dou mean that it needs an explicit entry in the dialplan.xml like: TEMPLATE MATCH=# Timeout=0 User=Phone/ !--Explicit # for Asterisk --Mike- Original Message - From: Andrew Latham [EMAIL PROTECTED]To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.comSent: Thursday, June 16, 2005 2:53 PMSubject: Re: [Asterisk-Users] Cisco 7960 (SIP) with Asterisk: how to get #towork during a call# and * are mapped later in the SIP(Default/MAC).cnf it has a section in the manual if you want to see why.On 6/16/05, Michael J. Tubby B.Sc (Hons) G8TIC [EMAIL PROTECTED]wrote: Gents, I've built an Asterisk system to replace our PBX at work and have Cisco 7960 phones (SIP 7.4) running with Asterisk 1.0.7. How to I get Asterisk to recognise the '#' being pressed during a call? In sip.conf I have entries likle this: [2001] type=friend context=local-phone auth=md5 username=2001 secret=xyzzy callerid=Jack Tubby 2001 host=dynamic nat=no canreinvite=no dtmfmode=rfc2833 incominglimit=2 [EMAIL PROTECTED] disallow=all allow=alaw allow=ulaw callgroup=2 pickupgroup=2 and in the SIPDefault.cnf for the phones I have: # Inband DTMF Settings (0-disable, 1-enable (default)) dtmf_inband: 1 # Out of band DTMF Settings (none-disable, avt-avt enable (default), avt_always - always avt ) dtmf_outofband: avt # DTMF dB Level Settings (1-6dB down, 2-3db down, 3-nominal (default), 4-3db up, 5-6dB up) dtmf_db_level: 3 DTMF works for voicemail and for remote services over both analogue Zap channels and digital (ISDN) channels. Asterisk doesn't appear to be 'monitoring' the audio so I can't get to Asterisk features like Asterisk's transfer, parked calls and one-tuch-record... Am I missing something? Mike ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --sigAndrew Latham - AKA: LATHAMA (lay-th-ham-eh)WWW: http://lathama.comEmail: [EMAIL PROTECTED] - [EMAIL PROTECTED] - [EMAIL PROTECTED]If any of the above are down we have bigger problems than my email!/sig___ Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users___ Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Analyzer for Milliwatt
Andrew Kohlsmith wrote: What is being discussed here is basically what I was planning on doing for an automatic VOIP quality check. Using miliwatt and analyzing it for pop/jitter/etc as well as sending other known waveforms and comparing what was received to what was expected and coming up with some quality number which would be fed back to the dialplan to adjust the least-cost routing paths. Essentially come up with a least cost but still good quality routing. :-) I've done absolutely nothing other than a little research and a lot of thinking about how to do it though. I did some research on digital click/pop removal for records as a way to detect poor quality, and then also some monkeying around with coppice's excellent DSP routines in spandsp. -A. Andrew, This sounds like a programming project. Something like a stripped down softphone (or possibly a plugin to an existing phone) with the ability to analyze the Milliwatt signal for variations/quality problems. The ability to analyze other known waveforms would add a lot of value. I suggest proposing your ideas to the -dev list or #asterisk-dev on FreeNode. Someone else (I can't recall who) is working with SIPP in order to get it to pass the full RTP stream, instead of just the SIP signaling. I believe that analyzing the quality of the RTP stream is still an open issue. If it could be handled on a 1-to-1 basis by the call endpoints, it sounds like an elegant and scalable solution. Currently, testing the scalability of an Asterisk system is a bit of a black art. We did some work with an Abacus 5000 http://www.spirentcom.com/analysis/product_set.cfm?PS=73PL=34wt=2, but they have a couple of significant drawbacks. It was capable of originating and terminating hundreds of SIP calls, but it could only do audio quality analysis on up to 64 of them. It is also a VERY expensive piece of equipment. I'm very interested in your project, because our production system will push the vertical scalability of Asterisk. So far we've handled 100 concurrent calls with digital recording on a single server in a live environment with no quality issues, but the number of calls is going to increase to the 400-500 range as we add clients to the box. The ability to test the results of the increased number of calls prior to going live could save me a LOT of headaches. As such, your project is of significant value to myself as well as the community at large. Please pursue it with the development community, and don't hesitate to contact me if needed. Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GPS-enabled cell phone/PDA
Date: Fri, 24 Feb 2006 14:56:54 + From: Steve Kennedy [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] GPS-enabled cell phone/PDA On Fri, Feb 24, 2006 at 07:17:52AM -0600, Rich Adamson wrote: Its my understanding the cell phone coordinates are sent to the cell phone provider and their equipment reads (and holds) that data. Its not part of any data available to you in any form unless you talk to the cell provider and convience them you have a valid need. Highly unlikely in the US anyway. Even if you could convience them to provide it, they would likely demaand some sort of out-of-band data transmission facility. GSM networks have the Cell ID available to the phone, however that's not much use without the location of the cellsite. There are now location based services, whereby you can query the network and they'll give out an approximate location (most cells are sectored [6 sectors per cell) which gives a direction, the cell also knows what power the phone is transmitting with, and the power it's received so can make a good approximation of where the phone is (within 60 degrees angle). However it's likely a phone will be picked up by several cells, so the network can triangulate and make a better aproximation. Making the information available to end-users is problematic due to privacy issues, unless the user explicitly agrees to give the info away. With GPS units, the info is stored in the phone and can send it out using SMS or other means. - It was my impression that only a handful of cellphones have full GPS units in them. Benefon and some Motorola units made for the former Nextel come to mind. The Benefon units do send SMS reports, and in fact, I have written code to control and track these units via SMS using a Nokia 31 GSM terminal. Unfortunately, aside from their unique GPS/SMS capability, the Benefons are not very attractive products, in my opinion. And they are expensive. The Motorola units contain Java machines and a well defined API for accessing the location data. I have not worked with them. There have undoubtedly been changes in the marketplace since I did this work about 2 years ago. As I understand it (but don't have thorough knowledge and could be mistaken), other units generally only receive GPS satellite signals and relay the data to cellular provider networks where the actual location calculation is done. This can be done with assistance of data obtained based on tower proximity, which jumpstarts the iterative process of approximation. I think it is called assisted GPS or some such... ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Trouble Chan Spy
Hi list, I got a question: When I try to ChanSpy a SIP channel I only listen one channel, for example, I call from 302 extension and I have two active channels: SIP/r1-voip-1b7b (None) Up Bridged Call(SIP/302-f1f1) SIP/302-f1f1 [EMAIL PROTECTED] Up Dial(SIP/[EMAIL PROTECTED]|4 When I try to spy this call from another extension: 1.SIP/301-fecc [EMAIL PROTECTED] Up ChanSpy(SIP/302) 2.SIP/r1-voip-1b7b (None) Up Bridged Call(SIP/302-f1f1) 3.SIP/302-f1f1 [EMAIL PROTECTED] Up Dial(SIP/[EMAIL PROTECTED]|4 I got 3 active channels, the one spying, the one that places the call and the one that receives the call. My problem is in the spying channel I can only hear the one that receives the call (3) but I cannot hear the channel (2): Thanks for your help, ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] fax receive using TDM400P
Anton Krall wrote: Well, I have the same effect on my TDM as in the E1 using unicall... Faxes get here as garbage :( I really would like to see sometime some audio recordings made by IAXmodem for people that had problems with TDMs and faxing with rxfax/txfax. Not that I have some hope of IAXmodem overcoming the odds, but because I'd like to actually see the what the TDM is doing to the audio. Lee. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] fax receive using TDM400P
Anton Krall wrote: Any modification made to zapata as far as echo and gains? As a rule, don't let anything manipulate the audio at all... even echo cancellation. That said, I have seen situations where gain had to be increased. Should echocancel be on or off? Off, most definitely off. I can't imagine an echo cancellor being capable of knowing what is echo and what isn't echo in a fax call. Lee. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] fax receive using TDM400P
Am Friday 24 February 2006 16:48 schrieb Anton Krall: Any modification made to zapata as far as echo and gains? Should echocancel be on or off? i have echocancel switched on, faxdetect is on, rx- and txgain is not used. (commented out). my var/log/messages says: Found a Wildcard TDM: Wildcard TDM400P REV I (4 modules) ... Zaptel Version: 1.2.4 Echo Canceller: KB1 maybe it depends on different hardware revisions? i don't know... tom |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Thomas Artner |Sent: Friday, February 24, 2006 8:25 AM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] fax receive using TDM400P | |Hi! | |I am using tdm400 cards for receiving faxes. It worked quite |out of the box. I installed spandsp for the rxfax application only. | |I use it as phone/fax switch: |All incoming calls are answered automatically to listen |whether its a fax or not. If it is a fax, the call is |forwarded to the buil-in fax extension, otherwise the analog |phones (all on tdm400) rings. | |It works without problems. Its for a small company (about a |few faxes per |hour) | | |Tom | |Am Freitag, 24. Februar 2006 07:10 schrieb Anton Krall: | Guys. | | Ive been testing how to receive faxes using TDM400P cards | |and so far, after | | playing with gains, echocancell and echotraining on | |zapata.conf.. Ive ha | | dno luck, faxes come in as garbage or broken or with blank lines. | | Anybody has successfully done this? Any tips.. Also I have | |some ideas: | 1. Is it really possible to get fxes on a fax machine using | |ATAs like the | | sipura 2002? Even using ulaw and pass-thru, is it possible? | | 2. Since the faxes is coming from PSTN thru the card, I | |guess asterisk will | | always stay in the middle right? No way around this. | | 3. Im also trying to receive faxes usign a TE110P card with spandsp, | unicall and E1 R2MFC, no luck also, some stuff, garbage and | |broken faxes. | | Anybody done this sucessfuly? | | Hope anybody can share their thoughts and insight on this. | | ___ | --Bandwidth and Colocation provided by Easynews.com -- | | Asterisk-Users mailing list | To UNSUBSCRIBE or update options visit: |http://lists.digium.com/mailman/listinfo/asterisk-users | |-- |Thomas Artner |___ |--Bandwidth and Colocation provided by Easynews.com -- | |Asterisk-Users mailing list |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] What business IP phone to use
Anything under 1ms is so far below the threshold of perceivable sound quality, echo, delay etc. that it's a mute point to discuss IMHO. Not even in any cumulative effect it may have. I can certainly see the advantages of SNMP for remote troubleshooting but hard to justify for small offices with less than 10 extensions. A good quality unmanaged switch is all you need IMHO. Not a cheap plastic Dlink or Linksys you buy at your local wallmart mind you. -Original Message- From: Conrad Wood [mailto:[EMAIL PROTECTED] Sent: Friday, February 24, 2006 3:02 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] What business IP phone to use On Fri, 2006-02-24 at 10:54 +1100, David Ankers wrote: Are you sure those switch figures are right? 16ms delay in the switch path sounds a bit long. Cisco's mid-range switches like the 2950 have switching times measured in micro seconds. Then again a 2626 procurve is only around $700. I meant micro-seconds, yes - my apologies. The 26xx series are ok, but I had specifically the 4108 in mind when I said 'good experience'. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with T1 installation
Brian Roy wrote: Not totally true. A PRI is 23b 1d. A DS1 (US) is a 24 channel circuit. Nitin - When you stop/start asterisk does it load all 24 channels? Any errors? How about zap show channel 1 in the CLI? Learn something new every day. Doug ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicemail 0 for operator call routing
Paul Tinsley wrote: Does anyone know of a way to specify what extension is dialed when 0 is pressed in the voicemail system. I have a situation where there is more than one secretary and they want the 0 to redirect to the appropriate secretary for the two groups of people. So an example would be: 555-1234 - voicemail - Secretary 1 555-1235 - voicemail - Secretary 2 Any help would be greatly appreciated. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users You can set up a db value for each extension as to what secretary group they belong to. When someone 0's out, have the secretary key looked up and then dialed, if no value is found have it dial a default secretary. Bruce ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call quality problems
I'm having difficulty with an Asterisk system. The external party has very good call quality, but the internal party hears clipping and drop outs. The WAN comes in from the Cisco IAD and into a LAN switch (DLink DGS-1005D w/ 802.1p) where the two public IPs are switched to different devices. One device is a FireBox device controlling a separate LAN with VPNs. The other device is eth0 on the Asterisk system. On the Asterisk eth1 is a 3Com 2226 LAN switch which connects Polycom IP501 phones. There are no PCs on this voice LAN. All ports on all LAN switches indicate full duplex. The quality problem doesn't appear to be volume related (a single call still has problems). The Polycom IP501s use SIP to the PBX, and the PBX uses SIP to the provider. The normal WWV time signal consists of a constant tone that is interrupted every second by a click. On the Polycom, each click can be heard, the tone starts, but the tone is clipped and there is silence until the next click. I've verified that QoS is enabled in the IAD. I would appreciate your thoughts. Thanks, -- Michael Welter Telecom Matters Corp. Denver, Colorado US +1.303.414.4980 [EMAIL PROTECTED] www.TelecomMatters.net ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] incoming peer register problem
Hi, i have several incoming sip peers (mostly ciscos) , with 1.0 i always registered them like this: register = @prepago-in [prepago-in] type=friend host=192.168.10.120 context = from-external dtmfmode=rfc2833 insecure=very ; required for incoming FWD calls Now with 1.2.4 it doesnt work any more, this is what i see in the CLI console Feb 24 11:40:18 WARNING[11142]: chan_sip.c:3207 sip_register: Format for registration is user[:secret[:[EMAIL PROTECTED]:port][/contact] at line 154 i dont need a user and pass in the ciscos, what should i put for user? thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GPS-enabled cell phone/PDA
Some more recent phones have the possibility to be connected to seperate GSM-boxes. E.g. there is a plug-in for the (older) Nokia 9210(i)/9290(i) Communicators and most of the Symbian phones with Bluetooth support can be connected to any Bluetooth-enabled GPS-mouse ... I think, getting the position data with a defined accuracy is not the problem. I'm quite satisfied with the location delivered by the CB channels of the base stations. Crucial is indeed, what kind of location based service you want to build and how the data gets to the server ... With flatrate contracts regarding SMS or GPRS-data it's not even a real question of costs anymore ... But we slowly are getting completely OT for ASTERISK ;-) ... For more info about context awareness and location based services probably take some time to read what some colleagues here are doing in research http://www.ist-mobilife.org/ -Jürgen Its my understanding the cell phone coordinates are sent to the cell phone provider and their equipment reads (and holds) that data. Its not part of any data available to you in any form unless you talk to the cell provider and convience them you have a valid need. Highly unlikely in the US anyway. Even if you could convience them to provide it, they would likely demaand some sort of out-of-band data transmission facility. GSM networks have the Cell ID available to the phone, however that's not much use without the location of the cellsite. There are now location based services, whereby you can query the network and they'll give out an approximate location (most cells are sectored [6 sectors per cell) which gives a direction, the cell also knows what power the phone is transmitting with, and the power it's received so can make a good approximation of where the phone is (within 60 degrees angle). However it's likely a phone will be picked up by several cells, so the network can triangulate and make a better aproximation. Making the information available to end-users is problematic due to privacy issues, unless the user explicitly agrees to give the info away. With GPS units, the info is stored in the phone and can send it out using SMS or other means. - It was my impression that only a handful of cellphones have full GPS units in them. Benefon and some Motorola units made for the former Nextel come to mind. The Benefon units do send SMS reports, and in fact, I have written code to control and track these units via SMS using a Nokia 31 GSM terminal. Unfortunately, aside from their unique GPS/SMS capability, the Benefons are not very attractive products, in my opinion. And they are expensive. The Motorola units contain Java machines and a well defined API for accessing the location data. I have not worked with them. There have undoubtedly been changes in the marketplace since I did this work about 2 years ago. As I understand it (but don't have thorough knowledge and could be mistaken), other units generally only receive GPS satellite signals and relay the data to cellular provider networks where the actual location calculation is done. This can be done with assistance of data obtained based on tower proximity, which jumpstarts the iterative process of approximation. I think it is called assisted GPS or some such... ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with T1 installation
Are you sure you're supposed to be using EM?
On Feb 24, 2006, at 5:39 AM, Nitin Joshi wrote:
Hi All,
I have installed a Digium TE110P card on an Asterisk 1.2.1 system. Its
connected directly to the PSTN. But I am unable to make outbound calls
on the zap channels. The light on the card is green. Asterisk CLI
shows all 24 channels when I give the command 'zap show channels'. I
also noticed that Asterisk CLI shows an incoming call every few
seconds on the 24th channel. This must be some kind of a timing
signal. This is he first time I am configuring a T1 so I must have
done something wrong I guess.
These are the commands I used to load the zap module:
modprobe zaptel
modprobe wcte11xp
ztcfg -vvv
---
my zaptel.conf is as follows:
span=1,1,0,esf,b8zs
em=1-24
loadzone = us
defaultzone=us
--
the zapata.conf is as follows:
[trunkgroups]
[channels]
group=1
language=en
signalling=em_w
usecallerid=yes
callerid=asreceived
context=default
echocancel=64
echocancelwhenbridged=yes
rxgain=1.0
txgain=1.0
channel = 1-2
group=2
language=en
signalling=em_w
usecallerid=yes
callerid=asreceived
context=default
echocancel=64
echocancelwhenbridged=yes
rxgain=1.0
txgain=1.0
channel = 3-24
--
In extensions.conf i have specified the following line:
[default]
exten = _ZX,1,Dial(zap/g1/${EXTEN},15,tr)
--
When I try to dial using the T1 line I get the following error :
Feb 24 06:56:53 NOTICE[5724]: app_dial.c:1010 dial_exec_full: Unable
to create channel of type 'Zap' (cause 0 - Unknown)
== Everyone is busy/congested at this time (1:0/0/1)
== Auto fallthrough, channel 'SIP/7180-a103' status is 'CHANUNAVAIL'
Any ideas guys?
Thanks and regards,
Nitin Joshi.
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Re: [Asterisk-Users] Is setting the variable _TRANSFER_CONTEXT required in features.conf?
Hi, Okay but then how do you transfer across contexts then? Thanks Moises Silva wrote: you need to set a TRANSFER_CONTEXT, either for the transferer or transferee channel. I dont know why, but res_features give priority to the transferee TRANSFER_CONTEXT, if not found, then use the transferer TRANSFER_CONTEXT. That context is used to match the extension to dial. So you can set this var to any context you want. Regards On 2/23/06, *Chuck Bunn* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi, Is setting the variable _TRANSFER_CONTEXT required in features.conf for Asterisk 1.2.4? It appears from the Wiki that transfers across contexts are not possible when this is set. If it is not set can one trasfer across contexts?? Thanks ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.375 / Virus Database: 268.1.0/269 - Release Date: 2/24/2006 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] What business IP phone to use
Aha, micro seconds in networking terms is normally written usecs or us (actually it's the greek letter mu as in ulaw) rather than ms which are milliseconds seconds - what had me puzzled was that it was stated that this could harm the voice path! The difference can also cause unnecessary delays and therefor echo in the path. For example, procurve switches typically have 13ms switching time, the high-end netgears about 21ms. As soon as you stack a couple of switches you are talking 26ms vs 42ms extra delay in the path! There is then only 8 usecs between the two switches, how on earth would this make any difference to the voice path at all? Let alone induce any echo... Obviously the originally poster didn't understand the difference. And based on this, he's probably advising people not to use Netgear switches for voice, oh dear. I'll jump in here to make a couple of comments relative to ethernet switches. Not all switches are created equal!!! If you take the cover off a switch, write down the part numbers for the chips used, and read the doc on those chips, you'll see major differences. (We've actually tested several switches over the past several years in real customer's networks as well.) Many entry level switches on the market have only minimal buffering for inbound and outbound packets. Its not uncommon for output buffers to be limited to one or two packets, and as a user, you can't chnage it. Port congestion frequently shows up when two (or more) devices connected to a switch (assume 100 mbs for now) try to communicate via a single upstream port (assume 100 mbs for now). The instantanous offered traffic is essentially 200 mbs, and the switch is expected to send that traffic out via a 100 mbs port. For those devices with minimal buffering, packets will be dropped. For newer switches with deeper buffers, some packets will be held up in the chip's internal queue waiting to get on the outbound port's wire. The delay in the buffer will become jitter, and depending upon exactly how many ports are contending for the outboud port, the jitter _can_ become noticable. (That _is_ one of the reasons why some switch vendors support QoS.) One can talk about wire speed throughput, etc, and it doesn't mean squat. Those are all marketing and sales words, not engineering specs. There are plenty of very well known switch vendors that purchase switches from other manufacturers and put their names on the front covers. Some of those have characteristics as noted above, while others manage the buffering and queuing much better then what their marketing/sales words imply. Its fairly common to see engineers in large corporate networks using workgroup switches to consolidate traffic from multiple wiring closets, and not pay any attention whatsoever to dropped packets in the switches. That's about the time when senior mgmt intervens and asks an external company to assess their network performance to resolve the internal fingerpointing. Our company has completed many of these. The _only_ way to know for sure what a switch is doing (eg, dropping pkts) is to ensure the switches have some form of network management. Even the simple Dell 2708 (eight port gig switch for $100) has some level of mgmt in it. Certainly not the best, but at least you can identify some issues. With the pricing drops that we've all seen over the last couple of years, its fairly easy to find managed switches at very reasonable cost. I'd _never_ using unmanaged switches in any environment where critical application data flows across the net, and I'd suggest voip traffic represents critical traffic in all production networks. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Is setting the variable _TRANSFER_CONTEXT required in features.conf?
it seems im not undestanding your question then. Could you provide a practical example?On 2/24/06, Chuck Bunn [EMAIL PROTECTED] wrote:Hi,Okay but then how do you transfer across contexts then? ThanksMoises Silva wrote: you need to set a TRANSFER_CONTEXT, either for the transferer or transferee channel. I dont know why, but res_features give priority to the transferee TRANSFER_CONTEXT, if not found, then use the transferer TRANSFER_CONTEXT. That context is used to match the extension to dial. So you can set this var to any context you want. Regards On 2/23/06, *Chuck Bunn* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi, Is setting the variable _TRANSFER_CONTEXT required in features.conf for Asterisk 1.2.4? It appears from the Wiki that transfers across contexts are not possible when this is set. If it is not set can one trasfer across contexts?? Thanks ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message.Checked by AVG Free Edition.Version: 7.1.375 / Virus Database: 268.1.0/269 - Release Date: 2/24/2006___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Missing 31 DTMF tones over ZAP
Hello, I'm posting this to the list in case others run into the same issue. I've recently been connecting * to a legacy Avaya InDEX switch over E1 ISDN PRI here in the UK. Everything was working OK, except that DTMF digits were not being recognised by * when sent by the Avaya switch to the * system. Instead, the background noise of the call centre would be silenced while users hit the keys on their phones - echo tests and RecordFile produced a flatline response. I had at first thought that the Avaya switch may not be sending them, however this was working when * was not in the call path. With further testing, I've found out that it is in fact only the first 31 DTMF tones that are missing - those following are picked up OK. I've got no idea why this should happen, and have kludged a fix by having the Avaya switch send out 31 'fake' tones before the user starts entering data (using Translation inside Route List). If anyone has come across this before and knows of a 'proper' fix, or even what might be causing the issue, I'd be very grateful for the information. Hope this helps, Matt King, M.A. Oxon. Managing Director, Orderly Software Ltd. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] problems with dialing
Hi, We're having problems dialing out to Asterisk from our Grandstream GXP-200 phones. About 2 of 3 times, when we dial, nothing happens. Looking at the console in max debug mode, there are no messages except the following: Feb 24 10:29:20 WARNING[2475]: chan_sip.c:1208 retrans_pkt: Maximum retries exceeded on transmission 9913b47bcd7[EMAIL PROTECTED] for seqno 4524 (Critical Response) Note: Early dial is set to Yes. DTMF is via SIP info. The phones are connected via a wireless bridge, range extender, and router to the asterisk box. Pinging the phone from the Asterisk box reveals a fairly long latency: 64 bytes from 192.168.10.100: icmp_seq=1 ttl=250 time=1110 ms64 bytes from 192.168.10.100: icmp_seq=2 ttl=250 time=114 ms64 bytes from 192.168.10.100: icmp_seq=3 ttl=250 time=21.8 ms64 bytes from 192.168.10.100: icmp_seq=4 ttl=250 time=33.4 ms64 bytes from 192.168.10.100: icmp_seq=5 ttl=250 time=4.46 ms64 bytes from 192.168.10.100: icmp_seq=6 ttl=250 time=57.4 ms Could this be the source of the problem? If so, would appreciate tips on how to work around this. Thanks in advance, WILL ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] disallow, allow codes
Hi, On the general section of my sip.conf I had a disallow=all. Then I put disallow=all, allow=g729, allow=ulaw on my users. It didn't work until I removed the disallow=all from the header. I know disallow=all in the header is totally useless in this case (since I have it for every user), but anyway, is this the correct behavior? Thank you Dov ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Is setting the variable _TRANSFER_CONTEXT required in features.conf?
Hi, I support multiple context on one asterisk server. I have a situation where there is a spa that has seperate voicemail and extensions and a resturant on the same campus that has different extensions and voicemail. They both use the same asterisk server but I do need the ability to transfer a caller from the spa to the resturant and vise versa. There are seperate phone lines comming in for the spa and resturant as well. Thanks Moises Silva wrote: it seems im not undestanding your question then. Could you provide a practical example? On 2/24/06, *Chuck Bunn* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi, Okay but then how do you transfer across contexts then? Thanks Moises Silva wrote: you need to set a TRANSFER_CONTEXT, either for the transferer or transferee channel. I dont know why, but res_features give priority to the transferee TRANSFER_CONTEXT, if not found, then use the transferer TRANSFER_CONTEXT. That context is used to match the extension to dial. So you can set this var to any context you want. Regards On 2/23/06, *Chuck Bunn* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi, Is setting the variable _TRANSFER_CONTEXT required in features.conf for Asterisk 1.2.4? It appears from the Wiki that transfers across contexts are not possible when this is set. If it is not set can one trasfer across contexts?? Thanks ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com http://Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.375 / Virus Database: 268.1.0/269 - Release Date: 2/24/2006 ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org http://www.gnu.org ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.375 / Virus Database: 268.1.0/269 - Release Date: 2/24/2006 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Missing 31 DTMF tones over ZAP
what zap device are you using? IIRC disalbing the vpmdtmf on a 406 or 411 might help you. I think it's done in wctxx4p.c On 2/24/06, Matt King [EMAIL PROTECTED] wrote: Hello, I'm posting this to the list in case others run into the same issue. I've recently been connecting * to a legacy Avaya InDEX switch over E1 ISDN PRI here in the UK. Everything was working OK, except that DTMF digits were not being recognised by * when sent by the Avaya switch to the * system. Instead, the background noise of the call centre would be silenced while users hit the keys on their phones - echo tests and RecordFile produced a flatline response. I had at first thought that the Avaya switch may not be sending them, however this was working when * was not in the call path. With further testing, I've found out that it is in fact only the first 31 DTMF tones that are missing - those following are picked up OK. I've got no idea why this should happen, and have kludged a fix by having the Avaya switch send out 31 'fake' tones before the user starts entering data (using Translation inside Route List). If anyone has come across this before and knows of a 'proper' fix, or even what might be causing the issue, I'd be very grateful for the information. Hope this helps, Matt King, M.A. Oxon. Managing Director, Orderly Software Ltd. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ImportVar Syntax
I am trying to use ImportVar to get some information out of a SIP/ZAP channel. I cannot seem to find an example of the syntax, or what variables I can access. Basically, I would like to output which person is being called. i.e: SIP/25 calls SIP/21. 25 executes a macro, and the result is SIP/21. The info that I want is stored in the channel's Direct Bridge variable. I have tried: ImportVar(TEST=SIP/25-6d2a|name) which doesn't seem to do anything. Looking through the code, the thing that I am looking for is: c-_bridge-name (in function handle_showchan). The voip-info page for ImportVar returns an error, and I couldn't find any occurance of ImportVar, except in pbx.c. Thanks in advance! Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: [Asterisk-Users ] RE: Monitor a call in progress. (Steve Totaro)
Steve, You wrote this referring to monitoring a call in Asterisk, how about from an IP phones LCD display screen: 1. go to www.google.com 2. type asterisk monitor application 3. click on the first result 4. read and implement 5. google is your friend I hope I made myself clear too ;-P Moreover, which phone can we use? We have a call shop cashier attended feature for call shops, but still need to display the call to the booth user... Regards, Max Glucksmann e-mail: [EMAIL PROTECTED] Web: http://www.comtel-networks.com Venezuela Teléfono: (0500) MAXITEL ext. 1011001 Fax: (0212) 953-0769 USA Phone: 1 (877) 467-2877 ext. 1011001 Fax: (954) 671-6800 BEGIN:VCARD VERSION:2.1 N:Glucksmann;Max FN:Max Glucksmann (Fax del trabajo) ORG:ComTel Networks, Corp. TITLE:Director TEL;WORK;VOICE:+1 (877) 467-2877 TEL;HOME;VOICE:+58 (500) MAXITEL (629-4835) TEL;CELL;VOICE:+58 (414) 250-0909 TEL;WORK;FAX:+1 (954) 671-6800 TEL;HOME;FAX:+58 (212) 285-3320 ADR;WORK:;;Aerocav 1614, PO Box 25304;Miami;FL.;33102-5304;Estados Unidos de América LABEL;WORK;ENCODING=QUOTED-PRINTABLE:Aerocav 1614, PO Box 25304=0D=0AMiami, FL. 33102-5304=0D=0AEstados Unidos de= Am=E9rica EMAIL;PREF;FAX:Max Glucksmann ([EMAIL PROTECTED])@+1 (954) 671-6800 REV:20051212T222729Z END:VCARD ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
