[asterisk-users] running twice
Dear I am using an asterisk 1.2.7.1 , with postgres and safe_Asterisk, for running, asterisk. but there is a problem, after 2-3 hours after restarting any things, top shows me, that, two asterisk, are now running, and one of them, gets 99.7 percent of cpu. Do you have any idea? Best Mani Check out the hottest 2008 models today at Yahoo! Autos. http://autos.yahoo.com/new_cars.html ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] running twice
show us the output of ur top command Pezhman Lali wrote: Dear I am using an asterisk 1.2.7.1 , with postgres and safe_Asterisk, for running, asterisk. but there is a problem, after 2-3 hours after restarting any things, top shows me, that, two asterisk, are now running, and one of them, gets 99.7 percent of cpu. Do you have any idea? Best Mani Check out the hottest 2008 models today at Yahoo! Autos. http://autos.yahoo.com/new_cars.html ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users EMAIL DISCLAIMER : This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. Any unauthorised distribution or copying is strictly prohibited. If you receive this transmission in error, please notify the sender by reply email and then destroy the message. Opinions, conclusions and other information in this message that do not relate to official business of Mascon shall be understood to be neither given nor endorsed by Mascon. Any information contained in this email, when addressed to Mascon clients is subject to the terms and conditions in governing client contract. Whilst Mascon takes steps to prevent the transmission of viruses via e-mail, we can not guarantee that any email or attachment is free from computer viruses and you are strongly advised to undertake your own anti-virus precautions. Mascon grants no warranties regarding performance, use or quality of any e-mail or attachment and undertakes no liability for loss or damage, howsoever caused. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] running twice
thanks Benjamin the folowing is the output of TOP. Best top - 08:23:09 up 15 days, 2:26, 2 users, load average: 1.31, 1.29, 1.24 Tasks: 109 total, 2 running, 107 sleeping, 0 stopped, 0 zombie Cpu(s): 97.0% us, 3.0% sy, 0.0% ni, 0.0% id, 0.0% wa, 0.0% hi, 0.0% si Mem:450456k total, 421384k used,29072k free, 94292k buffers Swap: 2096472k total, 88k used, 2096384k free, 202592k cached PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ COMMAND 2656 root 25 0 31248 5956 1060 R 94.2 1.3 6250:40 asterisk 16797 root 16 0 31972 14m 9804 S 1.1 3.3 37:38.23 asterisk 31104 root 16 0 40092 3760 3272 S 0.2 0.8 0:32.55 ser 15453 root 16 0 2128 1040 800 R 0.2 0.2 0:00.37 top 1 root 16 0 1992 660 568 S 0.0 0.1 0:19.79 init 2 root 34 19 000 S 0.0 0.0 0:00.00 ksoftirqd/0 3 root RT 0 000 S 0.0 0.0 0:00.00 watchdog/0 4 root 10 -5 000 S 0.0 0.0 0:00.00 events/0 5 root 10 -5 000 S 0.0 0.0 0:00.00 khelper 6 root 10 -5 000 S 0.0 0.0 0:00.00 kthread 8 root 10 -5 000 S 0.0 0.0 0:34.58 kblockd/0 9 root 20 -5 000 S 0.0 0.0 0:00.00 kacpid 78 root 10 -5 000 S 0.0 0.0 0:00.00 khubd 136 root 15 0 000 S 0.0 0.0 0:00.76 pdflush 137 root 15 0 000 S 0.0 0.0 0:00.55 pdflush 139 root 19 -5 000 S 0.0 0.0 0:00.00 aio/0 138 root 15 0 000 S 0.0 0.0 0:12.16 kswapd0 226 root 10 -5 000 S 0.0 0.0 0:00.00 kseriod 298 root 11 -5 000 S 0.0 0.0 0:00.00 kpsmoused 309 root 15 0 000 S 0.0 0.0 4:49.57 kjournald 376 root 14 -4 2204 684 384 S 0.0 0.2 0:00.21 udevd 859 root 11 -5 000 S 0.0 0.0 0:00.00 kmirrord 1193 root 16 0 1652 604 504 S 0.0 0.1 2:00.14 syslogd 1196 root 16 0 1604 400 332 S 0.0 0.1 0:00.01 klogd 1213 root 19 0 1596 468 396 S 0.0 0.1 0:00.00 acpid 1232 root 16 0 4972 1116 788 S 0.0 0.2 0:00.03 sshd 1252 root 15 0 2208 800 664 S 0.0 0.2 0:00.00 xinetd 1269 root 11 -5 000 S 0.0 0.0 0:00.00 kauditd 1305 postgres 16 0 12900 2800 2436 S 0.0 0.6 0:40.63 postmaster 1310 postgres 16 0 9960 632 264 S 0.0 0.1 0:00.36 postmaster 1319 postgres 15 0 13032 2272 1820 S 0.0 0.5 0:08.68 postmaster 1320 postgres 16 0 10960 1556 188 S 0.0 0.3 0:07.03 postmaster 1321 postgres 15 0 10152 720 272 S 0.0 0.2 0:05.35 postmaster 1365 root 16 0 23908 10m 6412 S 0.0 2.3 0:00.74 httpd 1374 root 16 0 5176 1204 656 S 0.0 0.3 0:27.56 crond 1386 apache16 0 31852 13m 3912 S 0.0 3.0 0:03.39 httpd 1387 apache16 0 31972 13m 3916 S 0.0 3.0 0:04.37 httpd 1388 apache16 0 31640 13m 3916 S 0.0 3.0 0:04.03 httpd 1389 apache16 0 30060 11m 3924 S 0.0 2.6 0:04.47 httpd 1390 apache16 0 31760 13m 3908 S 0.0 3.0 0:03.67 httpd 1393 apache16 0 31944 13m 3904 S 0.0 3.0 0:03.42 httpd 1394 apache16 0 30124 11m 3908 S 0.0 2.6 0:04.13 httpd 1395 apache16 0 31888 13m 3912 S 0.0 3.0 0:05.05 httpd 1396 root 16 0 2160 460 328 S 0.0 0.1 0:00.00 atd 1402 root 15 0 2748 952 840 S 0.0 0.2 14:55.32 server 1407 root 17 0 1584 408 356 S 0.0 0.1 0:00.00 mingetty 1408 root 17 0 1584 412 356 S 0.0 0.1 0:00.00 mingetty 1409 root 17 0 1584 412 356 S 0.0 0.1 0:00.00 mingetty 1410 root 17 0 1588 412 356 S 0.0 0.1 0:00.00 mingetty 1418 root 18 0 1584 412 356 S 0.0 0.1 0:00.00 mingetty 1421 root 18 0 1588 416 356 S 0.0 0.1 0:00.00 mingetty 2488 root 16 0 2532 432 352 S 0.0 0.1 0:01.51 rtpproxy 7933 root 16 0 6952 5036 1644 S 0.0 1.1 0:00.70 miniserv.pl 30459 apache16 0 29956 11m 3812 S 0.0 2.6 0:01.97 httpd 31092 root 20 0 40092 3324 2904 S 0.0 0.7 0:00.03 ser --- Benjamin Jacob [EMAIL PROTECTED] wrote: show us the output of ur top command Pezhman Lali wrote: Dear I am using an asterisk 1.2.7.1 , with postgres and safe_Asterisk, for running, asterisk. but there is a problem, after 2-3 hours after restarting any things, top shows me, that, two asterisk, are now running, and one of them, gets 99.7 percent of cpu. Do you have any idea? Best Mani Check out the hottest 2008 models today at Yahoo! Autos. http://autos.yahoo.com/new_cars.html ___ Sign up now for AstriCon 2007! September 25-28th.
Re: [asterisk-users] Completing my Configuration
Am Dienstag, den 25.09.2007, 07:37 +0200 schrieb Guenther Sohler:
Hallo Group,
I have basically set up a small asterisk system,
which ahs 4 peers:
* registers at 2 Sipgates
* 2 hardware phones connected to it
Both Hardware phones can phone outwards(cheaper sipgate is selected with
dialplan)
Calls from both sipgates make my hardware phones ring
But here comes the challenges:
Is it possible to configure asterisk in such a way that in the phone:
* there are names instead of numbers in my hardware phone displayed
Depends on the hardware phones. In theory, with each SIP call connecting
to the phone, both a name and a number can be transferred. AFAIK sipgate
defaults to setting both to the usual callerID. That is exactly the
reason why you can set the variables ${CALLERID(num)} and
${CALLERID(name)}.
Some hardware phones (I assume, the better ones ;-) display both; my
Allnet for example seems to only display the name, but store the number
for the call back list. My Fritz!Boxen seem to forward both name and
number to ISDN devices on the internal S0-bus, just not many ISDN phones
can actually display text numbers.
Let your asterisk have an ast database, looking like
callerid/420123456789 = Doe, John Q.
callerid/492240224922 = Mustermann, Dr. Peter
Then you could expand your dialplan logic a little. If you have a line
exten = 12345,4,Dial(SIP/phone1,60)
or whatever that looks like in your SIP-incoming context, insert those
lines before it [and change the 4, 5, 6, 7s ;-) ]
exten = 12345,4,Set(CALLERID(name)=${DB(callerid/${CALLERID(num)})})
exten = 12345,5,GotoIf($[${CALLERID(name)} = ]?6:7)
exten = 12345,6,Set(CALLERID(name)=-- ${CALLERID(num)})
exten = 12345,7,Dial(SIP/phone1,60)
Line 6 treats the case that the number is not in your database and sets
the callerid-name to -- NUMBER_OF_CALLER
You can manually add data to the astdb from the asterisk CLI with
database set callerid 420456789 Silly, Roger M.
You should check that both your SIP providers provide incoming CLI in
the international formatting, without country prefix or +. In my
experience some SIP providers send numbers like
492240224922, others send +49... or 0049..., some send national format
02240... for all national calls, some even omit the leading 0 there,
and some just change the behaviour depending from which network (T-Com
landline, Arcor landline, T-Mobile cell phone, O2 cellphone, foreign
callers...) the call originates. If you have more than two providers,
this can be a PITA - you will need some dialplan logic to sanitize the
callerid in those cases, and sometimes you are just left for guessing,
for example when the provider signals calls from T-Mobile as 16177554224
and calls from Boston, MA, USA the very same. Germany does not have
fixed-length numbers, even in the mobile phone networks the length
differs, and the number given might be valid for both circumstances.
/rant
* The Ringtone is different for special call numbers
If your phone supports that, yes, you can do it. The common method for
this seems to be sending an additional header. There will be docs on
SIPAddHeader(blah) or similar on www.voip-info.org, and you might want
to also use a database here to find out wether special ringtones are to
be activated or not.
* it is displayed, in which sipgate the call came from
You could use the CALLERID(name) field for that, by adding the provider
short name in front of the caller's name, like
exten = 12345,4,Set(CALLERID(name)=at-${DB(callerid/${CALLERID(num)})})
for calls via the at provider - or whatever seems stylish enough.
I personally have a logic that makes use of the dial-around prefix in
use here in Germany: From a regular T-Com landline you can select the
provider that will carry the next call by dialling 010[1-9]X or 0100XX.
Those prefixes of course do not work on SIP provider lines, and my
asterisk does not have landlines connected. So I use those for my own
purposes, e.g. selecting the SIP account that the call may go out
through. Dialplan logic detects 010XX (100 possible accounts are
enough, I just ignore 0100XX as additional number field here) and
selects the outgoing provider accordingly.
If I wished to have the incoming line signalled to me, I would prefix
the incoming CALLERID(num) with the provider code. Callbacks would go
through the same line - nice bonus. Most of my phones do not handle text
and number simultaneous display in a reasonable way, so I do not rely on
the text.
* using an extension in my call number redirects the call just to one
sip phone ?
AFAIK you could only do this by Answer()ing the line (at which point the
caller starts paying the connection) and asking the caller to input an
extension. (Hint: Read()). I personally do not like this solution at
all, because that is what DID and number block allocation were invented
for. You can get a number block with SIP from some providers. Or you
just get yourself another private phone number ;-)
BR,
Anselm
[asterisk-users] HOWTO/FAQ question (Location: Sweden)
Sorry for this. This is most likely a HOWTO or FAQ question, but it's so much information and documentation to wade through so I hope someone could take a minute to answer anyway. If not, no worries. I'll get to it sooner or later :) I'm trying to understand what Asterisk actually is and the basic workings... I think I've understand what I need to get going, except one thing. How do I connect to a 'normal' (i.e. analog) telephone? That is, if my company/project have 100% IP telephony, but one of these phones need to call a analog telephone in another company (or if I need to call home for any reason :). What do I need from the 'phone company'? And what hardware? This is Sweden with Telia as provider if that matters. DISCLAIMER1: I've read about BRI, PRI, E1/T1 etc, but the reason I'm all confused is that they (Telia) don't seem to understand the question, and also claims that a PRI == E1. DISCLAIMER2: I've seen the Digium cards, but due to confusion with Telia, I'm not sure if I want/need a Digital or an Analog card... And 'how big'... ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] [EVENT] Asterisk and VoIP in enterprise
Hi, I'm sending this mail here as I think it is of interest for at least europeans amongst you. The event presented below is not commercially-oriented, but really of interest for professional Asterisk users. in two weeks (910 october), the event Asterisk and VoIP in enterprise will take place in Brussels, Belgium. It is organised by Profoss ( http://www.profoss.eu ) to spread information about the possibility to use Asterisk in professional environments, and will feature talks based on real world experience by professionals, including Kevin P. Fleming, Asterisk's co-maintainer. The talks will cover case studies, the integration of a proprietary product with Asterisk, debunking Asterisk myths, how to improve customer service with Asterisk, and more A round table with closed source vendors of competing products (with amongst themCISCO and Avaya) will also be organised so participants can get a global view of the VoIP market. Talks won't be commercial shows! This event is for ICT professionals: end users, consultants or Asterisk solutions providers. We have sincerely worked hard to make this an interesting event! All information about this event is available at http://www.profoss.eu/events/october-2007-asterisk/ When you register, use the code DIGIUMML, you won't regret it ;-) Raph ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] HOWTO/FAQ question (Location: Sweden)
Turbo Fredriksson wrote: Sorry for this. This is most likely a HOWTO or FAQ question, but it's so much information and documentation to wade through so I hope someone could take a minute to answer anyway. If not, no worries. I'll get to it sooner or later :) I'm trying to understand what Asterisk actually is and the basic workings... I think I've understand what I need to get going, except one thing. How do I connect to a 'normal' (i.e. analog) telephone? That is, if my company/project have 100% IP telephony, but one of these phones need to call a analog telephone in another company (or if I need to call home for any reason :). What do I need from the 'phone company'? And what hardware? You need something to interconnect to your telco, this can be done on different ways: - you can take a voip provider and not buy any hardware. - you can use a TDM card and connect to a classic analog telephone line. (1 simultaneous call per port) - you can use a BRI card and connect to a classic isdn line (You get 2 lines per port this case) If you need more capacity, (more than 2 simultaneous channels), just think PRI or PRA or E1 as one and the same thing. (30 simultaneous calls per port). Zoa. This is Sweden with Telia as provider if that matters. DISCLAIMER1: I've read about BRI, PRI, E1/T1 etc, but the reason I'm all confused is that they (Telia) don't seem to understand the question, and also claims that a PRI == E1. DISCLAIMER2: I've seen the Digium cards, but due to confusion with Telia, I'm not sure if I want/need a Digital or an Analog card... And 'how big'... ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] HOWTO/FAQ question (Location: Sweden)
Hi, On 9/25/07, Turbo Fredriksson [EMAIL PROTECTED] wrote: Sorry for this. This is most likely a HOWTO or FAQ question, but it's so much information and documentation to wade through so I hope someone could take a minute to answer anyway. If not, no worries. I'll get to it sooner or later :) I'm trying to understand what Asterisk actually is and the basic workings... I think I've understand what I need to get going, except one thing. asterisk is a IP-PBX system you can visit asterisk.org for more info or voip-info.org. How do I connect to a 'normal' (i.e. analog) telephone? That is, if my company/project have 100% IP telephony, but one of these phones need to call a analog telephone in another company (or if I need to call home for any reason :). What do I need from the 'phone company'? And what hardware? This is Sweden with Telia as provider if that matters. IT depends with your provider, If they use analog TDM lines you'll be needing Digium TDM400P or TDM2400P with fxo modules, If they can do digital E1 PRI you can use Digium TE120P(one E1 port), TE207P/TE212P(two E1 port), TE407P/TE412P(four E1 port). Or for pci express cards either TE220P or TE420. you can check digium.org for more details of this cards. You can also use your analog phone devices either connect it through ATA(Analog Telephony Adaptor) example are Linksys PAP2, Digium TDM400P(4-port) with fxs module or TDM2400P (24-port) with fxs module. DISCLAIMER1: I've read about BRI, PRI, E1/T1 etc, but the reason I'm all confused is that they (Telia) don't seem to understand the question, and also claims that a PRI == E1. AFAIK PRI is the signalling for E1. DISCLAIMER2: I've seen the Digium cards, but due to confusion with Telia, I'm not sure if I want/need a Digital or an Analog card... And 'how big'... if they said they can do PRI you'll be needing a Digium card. but if they can use VOIP go with VOIP to save up from buying the hardware. HTH -- Regards, Mark Quitoriano, CCNA Fan the flame... http://www.spreadfirefox.com/?q=user/registerr=19441 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Grandstream GXP2020 / 2000
Hi, Has somebody experiences with the Grandstream GXP2020 / 2000 phones in a business graded installation (with really traffic on not 3 calls a day ;-) ) Of course with Asterisk PBX (1.4.1 or 1.4.11 or 1.4 in generall) Thanks! Kind Regards, Erik ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Backuping VoIP provider with PRI
Hi list,
My Asterisk config for outgoing calls is the following:
exten = s,1,Dial(SIP/[EMAIL PROTECTED],60,g)
exten = s,n,GotoIf($[\${ANSWEREDTIME}\ = \\]?pri:hang)
exten = s,n(pri),NoOp(Problems with voip provider trying PRI)
exten = s,n,Dial(Zap/g2/${MACRO_EXTEN},60,g)
exten = s,n(hang),HangUp
in most cases it works well but, if my internet connection is down
Asterisk tries to Dial voipprovider, but it can't resolve the dns name,
so it waits 60 seconds to jump to the following priority...
Any ideas to solve this problem? I can't use the IP of the provider (it
has a pool of servers), I try to use dnsmgr without solving the isue
Thanks in advance,
Marc
PD: I have used more sophisticate configs using DIALSTATUS variable, but
the problem persists
___
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[asterisk-users] Anyone else having problems with the list
I have sent a few emails over the past couple of days that simply have not arrived on the list (or so it seems). Is anyone else encountering this ? Julian ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Completing my Configuration
Dear Anselm,
I am sorry about the big traffic in the newsgroup.
I tried to send my post to the newsgroup for 3 days now - once a day,
but it did not appear. Today I tried putting it in cc also with, then it
worked out ...
I will carefully read your answer.
thank you veryy much
Original-Nachricht
Datum: Tue, 25 Sep 2007 09:34:14 +0200
Von: Anselm Martin Hoffmeister [EMAIL PROTECTED]
An: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Betreff: Re: [asterisk-users] Completing my Configuration
Am Dienstag, den 25.09.2007, 07:37 +0200 schrieb Guenther Sohler:
Hallo Group,
I have basically set up a small asterisk system,
which ahs 4 peers:
* registers at 2 Sipgates
* 2 hardware phones connected to it
Both Hardware phones can phone outwards(cheaper sipgate is selected with
dialplan)
Calls from both sipgates make my hardware phones ring
But here comes the challenges:
Is it possible to configure asterisk in such a way that in the phone:
* there are names instead of numbers in my hardware phone displayed
Depends on the hardware phones. In theory, with each SIP call connecting
to the phone, both a name and a number can be transferred. AFAIK sipgate
defaults to setting both to the usual callerID. That is exactly the
reason why you can set the variables ${CALLERID(num)} and
${CALLERID(name)}.
Some hardware phones (I assume, the better ones ;-) display both; my
Allnet for example seems to only display the name, but store the number
for the call back list. My Fritz!Boxen seem to forward both name and
number to ISDN devices on the internal S0-bus, just not many ISDN phones
can actually display text numbers.
Let your asterisk have an ast database, looking like
callerid/420123456789 = Doe, John Q.
callerid/492240224922 = Mustermann, Dr. Peter
Then you could expand your dialplan logic a little. If you have a line
exten = 12345,4,Dial(SIP/phone1,60)
or whatever that looks like in your SIP-incoming context, insert those
lines before it [and change the 4, 5, 6, 7s ;-) ]
exten = 12345,4,Set(CALLERID(name)=${DB(callerid/${CALLERID(num)})})
exten = 12345,5,GotoIf($[${CALLERID(name)} = ]?6:7)
exten = 12345,6,Set(CALLERID(name)=-- ${CALLERID(num)})
exten = 12345,7,Dial(SIP/phone1,60)
Line 6 treats the case that the number is not in your database and sets
the callerid-name to -- NUMBER_OF_CALLER
You can manually add data to the astdb from the asterisk CLI with
database set callerid 420456789 Silly, Roger M.
You should check that both your SIP providers provide incoming CLI in
the international formatting, without country prefix or +. In my
experience some SIP providers send numbers like
492240224922, others send +49... or 0049..., some send national format
02240... for all national calls, some even omit the leading 0 there,
and some just change the behaviour depending from which network (T-Com
landline, Arcor landline, T-Mobile cell phone, O2 cellphone, foreign
callers...) the call originates. If you have more than two providers,
this can be a PITA - you will need some dialplan logic to sanitize the
callerid in those cases, and sometimes you are just left for guessing,
for example when the provider signals calls from T-Mobile as 16177554224
and calls from Boston, MA, USA the very same. Germany does not have
fixed-length numbers, even in the mobile phone networks the length
differs, and the number given might be valid for both circumstances.
/rant
* The Ringtone is different for special call numbers
If your phone supports that, yes, you can do it. The common method for
this seems to be sending an additional header. There will be docs on
SIPAddHeader(blah) or similar on www.voip-info.org, and you might want
to also use a database here to find out wether special ringtones are to
be activated or not.
* it is displayed, in which sipgate the call came from
You could use the CALLERID(name) field for that, by adding the provider
short name in front of the caller's name, like
exten = 12345,4,Set(CALLERID(name)=at-${DB(callerid/${CALLERID(num)})})
for calls via the at provider - or whatever seems stylish enough.
I personally have a logic that makes use of the dial-around prefix in
use here in Germany: From a regular T-Com landline you can select the
provider that will carry the next call by dialling 010[1-9]X or 0100XX.
Those prefixes of course do not work on SIP provider lines, and my
asterisk does not have landlines connected. So I use those for my own
purposes, e.g. selecting the SIP account that the call may go out
through. Dialplan logic detects 010XX (100 possible accounts are
enough, I just ignore 0100XX as additional number field here) and
selects the outgoing provider accordingly.
If I wished to have the incoming line signalled to me, I would prefix
the incoming CALLERID(num) with the provider code. Callbacks would go
[asterisk-users] Asterisk Redundancy
Hi All, I'm interested in how people are clustering Asterisk, if that's possible, or how you might be achieving a redundant solution. I've a single Asterisk server driving the company. Its well backed-up, and I've a cloned machine that (in theory) with a DNS change could take over operations. However I'd like to achieve something more automated if possible. I know that some of my VoIP trunk providers cluster IAX connections, but I'm not sure how they would do that. Any ideas? Adrian Marsh ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Backuping VoIP provider with PRI
Marc Patino Gómez wrote: in most cases it works well but, if my internet connection is down Asterisk tries to Dial voipprovider, but it can't resolve the dns name, so it waits 60 seconds to jump to the following priority... Any ideas to solve this problem? I can't use the IP of the provider (it has a pool of servers), I try to use dnsmgr without solving the isue Why don't you fill the ip addresses to your /etc/hosts file? In that way lookups won't need any dns resolving and still could keep the load balancing by having multiple ip addresses to the same SIP hostname. regards Adam ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and OCS integration
Yes, I have read some articles about it. But I would like to try something similar to http://www.opensubscriber.com/message/asterisk-users@lists.digium.com/1939626.html Any experience in this? From: Jon Schøpzinsky [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk and OCS integration Date: Mon, 24 Sep 2007 16:03:40 +0200 I would use SER or OpenSER as a middle man. Set it up to receive via TCP and send it on to the asterisk server using UDP. Kind Regards Jon Leren Schøpzinsky Solution Engineer Dansk Erhvervs-Telefon A/S tlf: +45 88200336 mob: +45 31206709 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of dadsadsadf dsadasdsa Sent: 24. september 2007 13:29 To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk and OCS integration Hi List! does anyone played around with the OCS and Asterisk? I want to integrate OCS and Asterisk to enable Office Communicator 7.0 client to make and receive calls from PSTN I know that I need patch Asterisk to support TCP. But I am a bit ( a lot) lost Which more things should I need to keep in mind? Any advise will be wellcome :-) Thank you very much, Marta _ Acepta el reto MSN Premium: Protección para tus hijos en internet. Descárgalo y pruébalo 2 meses gratis. http://join.msn.com?XAPID=1697DI=1055HL=Footer_mailsenviados_proteccioninfantil ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Descarga gratis la Barra de Herramientas de MSN http://www.msn.es/usuario/busqueda/barra?XAPID=2031DI=1055SU=http%3A//www.hotmail.comHL=LINKTAG1OPENINGTEXT_MSNBH ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and OCS integration
Yes, I have read some articles about it. But I would like to try something similar to http://www.opensubscriber.com/message/asterisk-users@lists.digium.com/1939626.html Any experience in this? From: Jon Schøpzinsky [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk and OCS integration Date: Mon, 24 Sep 2007 16:03:40 +0200 I would use SER or OpenSER as a middle man. Set it up to receive via TCP and send it on to the asterisk server using UDP. Kind Regards Jon Leren Schøpzinsky Solution Engineer Dansk Erhvervs-Telefon A/S tlf: +45 88200336 mob: +45 31206709 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of dadsadsadf dsadasdsa Sent: 24. september 2007 13:29 To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk and OCS integration Hi List! does anyone played around with the OCS and Asterisk? I want to integrate OCS and Asterisk to enable Office Communicator 7.0 client to make and receive calls from PSTN I know that I need patch Asterisk to support TCP. But I am a bit ( a lot) lost Which more things should I need to keep in mind? Any advise will be wellcome :-) Thank you very much, Marta _ Acepta el reto MSN Premium: Protección para tus hijos en internet. Descárgalo y pruébalo 2 meses gratis. http://join.msn.com?XAPID=1697DI=1055HL=Footer_mailsenviados_proteccioninfantil ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Dale rienda suelta a tu tiempo libre. Mil ideas para exprimir tu ocio con MSN Entretenimiento. http://entretenimiento.msn.es/ ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anyone else having problems with the list
Yes for me. Carlos -- Julian Lyndon-Smith wrote: I have sent a few emails over the past couple of days that simply have not arrived on the list (or so it seems). Is anyone else encountering this ? Julian ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Help with Sip Registration
Hi all, I have installed X-lite client on a windowsXP machine and asterisk on an enterprise linux m/c. The client is sending a registration message to asterisk server. It is able to process the message and sends 200 OK back. But later it says Scheduling destruction of sip dialog . Then it says Really destroying sip dialog . What to do for this problem??? I had enabled the sip debug at the asterisk. I have pasted the messages, I got below. Please help me in solving the problem. Thanks in advance, Treesa REGISTER sip:192.168.12.160 SIP/2.0 Via: SIP/2.0/UDP 192.168.25.116:52166;branch=z9hG4bK-d87543-b65c86230550ff4f- 1--d87543-;rport Max-Forwards: 70 Contact: sip:[EMAIL PROTECTED]:52166;rinstance=1a12ef13351e0ee1 To: 1002sip:[EMAIL PROTECTED] From: 1002sip:[EMAIL PROTECTED];tag=5f799517 Call-ID: Yjk5ZTFjZjI1Mzc5N2RiMWFhYWRiMjhhMjc0NTU4ZDI. CSeq: 1 REGISTER Expires: 3600 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO User-Agent: X-Lite release 1011s stamp 41150 Content-Length: 0 - --- (12 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 192.168.25.116 : 52166 (NAT) localhost*CLI --- Transmitting (NAT) to 192.168.25.116:52166 --- SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.25.116:52166;branch=z9hG4bK-d87543-b65c86230550ff4f- 1--d87543-;received=192.168.25.116;rport=52166 From: 1002sip:[EMAIL PROTECTED];tag=5f799517 To: 1002sip:[EMAIL PROTECTED] Call-ID: Yjk5ZTFjZjI1Mzc5N2RiMWFhYWRiMjhhMjc0NTU4ZDI. CSeq: 1 REGISTER User-Agent: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: sip:[EMAIL PROTECTED] Content-Length: 0 localhost*CLI --- Transmitting (NAT) to 192.168.25.116:52166 --- SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.25.116:52166;branch=z9hG4bK-d87543-b65c86230550ff4f- 1--d87543-;received=192.168.25.116;rport=52166 From: 1002sip:[EMAIL PROTECTED];tag=5f799517 To: 1002sip:[EMAIL PROTECTED];tag=as13bc832b Call-ID: Yjk5ZTFjZjI1Mzc5N2RiMWFhYWRiMjhhMjc0NTU4ZDI. CSeq: 1 REGISTER User-Agent: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=09891096 Content-Length: 0 Scheduling destruction of SIP dialog 'Yjk5ZTFjZjI1Mzc5N2RiMWFhYWRiMjhhMjc0NTU4ZDI.' in 32000 ms (Method: REGISTER) localhost*CLI --- SIP read from 192.168.25.116:52166 --- REGISTER sip:192.168.12.160 SIP/2.0 Via: SIP/2.0/UDP 192.168.25.116:52166;branch=z9hG4bK-d87543-4d49b62ac70e0843- 1--d87543-;rport Max-Forwards: 70 Contact: sip:[EMAIL PROTECTED]:52166;rinstance=1a12ef13351e0ee1 To: 1002sip:[EMAIL PROTECTED] From: 1002sip:[EMAIL PROTECTED];tag=5f799517 Call-ID: Yjk5ZTFjZjI1Mzc5N2RiMWFhYWRiMjhhMjc0NTU4ZDI. CSeq: 2 REGISTER Expires: 3600 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO User-Agent: X-Lite release 1011s stamp 41150 Authorization: Digest username=1002,realm=asterisk,nonce=09891096,uri=sip:192.168.12 .160,response=f9d4e4b46f8da4d0c2fc7fe4e1f4c7fe,algorithm=MD5 Content-Length: 0 - --- (13 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 192.168.25.116 : 52166 (NAT) --- Transmitting (NAT) to 192.168.25.116:52166 --- SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.25.116:52166;branch=z9hG4bK-d87543-4d49b62ac70e0843- 1--d87543-;received=192.168.25.116;rport=52166 From: 1002sip:[EMAIL PROTECTED];tag=5f799517 To: 1002sip:[EMAIL PROTECTED] Call-ID: Yjk5ZTFjZjI1Mzc5N2RiMWFhYWRiMjhhMjc0NTU4ZDI. CSeq: 2 REGISTER User-Agent: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: sip:[EMAIL PROTECTED] Content-Length: 0 -- Registered SIP '1002' at 192.168.25.116 port 52166 expires 3600 -- Saved useragent X-Lite release 1011s stamp 41150 for peer 1002 --- Transmitting (NAT) to 192.168.25.116:52166 --- SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.25.116:52166;branch=z9hG4bK-d87543-4d49b62ac70e0843- 1--d87543-;received=192.168.25.116;rport=52166 From: 1002sip:[EMAIL PROTECTED];tag=5f799517 To: 1002sip:[EMAIL PROTECTED];tag=as13bc832b Call-ID: Yjk5ZTFjZjI1Mzc5N2RiMWFhYWRiMjhhMjc0NTU4ZDI. CSeq: 2 REGISTER User-Agent: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 3600 Contact: sip:[EMAIL PROTECTED]:52166;rinstance=1a12ef13351e0ee1;expires=3600 Date: Fri, 15 Aug 2014 16:49:53 GMT Content-Length: 0 Scheduling destruction of SIP dialog 'Yjk5ZTFjZjI1Mzc5N2RiMWFhYWRiMjhhMjc0NTU4ZDI.' in 32000 ms (Method: REGISTER) localhost*CLI --- SIP read from 192.168.25.116:52166 --- - --- (0 headers 1 lines) --- Really destroying SIP dialog 'Yjk5ZTFjZjI1Mzc5N2RiMWFhYWRiMjhhMjc0NTU4ZDI.' Method: REGISTER ___
Re: [asterisk-users] Anyone else having problems with the list
On 22:43, Tue 25 Sep 07, Carlos Hernandez wrote: Yes for me. Carlos -- Julian Lyndon-Smith wrote: I have sent a few emails over the past couple of days that simply have not arrived on the list (or so it seems). Is anyone else encountering this ? Julian I have similar problems. Some mails arrive, some dont. If I check the listarchive on the web I see more emails then in mutt. I already disabled greylisting etc and browsed thru the spam quarantine but nothing. -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called users? ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Backuping VoIP provider with PRI
Hi Adam, thanks for your quick answer, I try your tip but the problem persist, so... It seems not to be a dns problem Asterisk executes the Dial command and it tries to reach the VoIP provider until timeout, in * console appears: Called [EMAIL PROTECTED] Anybody knows howto make dial command don't wait until timeout when the provider host is unrechable? Cheers, Marc Adam KOSA wrote: Marc Patino Gómez wrote: in most cases it works well but, if my internet connection is down Asterisk tries to Dial voipprovider, but it can't resolve the dns name, so it waits 60 seconds to jump to the following priority... Any ideas to solve this problem? I can't use the IP of the provider (it has a pool of servers), I try to use dnsmgr without solving the isue Why don't you fill the ip addresses to your /etc/hosts file? In that way lookups won't need any dns resolving and still could keep the load balancing by having multiple ip addresses to the same SIP hostname. regards Adam ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Hola Jonathan, a ver si tre suena...
Hola Jonathan Te cuento un pokillo lo q intento hacer por si me puedes orientar en algo o de algun sitio donde pueda mirar Existe una especificación de Microsoft de lo que llaman Dual-Forking, que básicamente consiste en poder usar tanto el teléfono como el propio PC como dispositivo de comunicaciones, según convenga. De esta manera, por ejemplo, se puede usar estando en la oficina el teléfono de nuestra IP PBX para hacer y recibir llamadas, y sin embargo si se está de viaje, es posible usar el propio PC para iniciar y recibir llamadas a teléfonos IP de una empresa o incluso a RTC a través de su infraestructura de Voz IP. Para gente móvil o en general para perfiles de cliente que se inclinen por una solución softphone, este desarrollo es clave. Uno de los problemas a los q me enfrentaba para intentar evaluar la historia es q en Asterisk tenemos sip sobrre udp y MS funciona sobre tcp. No sé si tú me puedes orientar un pokillo enq mas cosas deberia tener en cuenta, o si conoces alguna experiencia/modelo similar , etc. Weno, pues un saludo y muchas gracias (aun no me dijiste de donde eres) Marta From: [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Subject: Re: [asterisk-users] SIP over TCP Date: Fri, 21 Sep 2007 21:05:59 +0200 SPANISH IOLLOWS...sorry about that folks! -- hola Marta, bienvenida al mundo de las PBX y Asterisk. Tambien enhorabuena en tu andadura profesional. Como se que al principio es dificil no me importaria echarte una mano para que despegue tu proyecto laboral :) Toma nota de mi email y escribeme sin compromiso. Un saludo. Jonathan GF On 9/21/07, dadsadsadf dsadasdsa [EMAIL PROTECTED] wrote: Hy all, I am Marta from Spain. I have just start working and my first project is with Asterisk. And I am a bit lost⦠I am interested in testing sip over tcp. I have read in http://www.sineapps.com/news.php?rssid=1777 that there are some implementations. Is this the most recent version? Are there any other developments in this area? Is it really working? How can I test it? Thank you very much, Marta _ Acepta el reto MSN Premium: Protección para tus hijos en internet. Descárgalo y pruébalo 2 meses gratis. http://join.msn.com?XAPID=1697DI=1055HL=Footer_mailsenviados_proteccioninfantil ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Descubre la descarga digital con MSN Music. Más de un millón de canciones. http://music.msn.es/ ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Backuping VoIP provider with PRI
Qualify=yes? Thanks, Steve Marc Patino Gómez wrote: Hi Adam, thanks for your quick answer, I try your tip but the problem persist, so... It seems not to be a dns problem Asterisk executes the Dial command and it tries to reach the VoIP provider until timeout, in * console appears: Called [EMAIL PROTECTED] Anybody knows howto make dial command don't wait until timeout when the provider host is unrechable? Cheers, Marc Adam KOSA wrote: Marc Patino Gómez wrote: in most cases it works well but, if my internet connection is down Asterisk tries to Dial voipprovider, but it can't resolve the dns name, so it waits 60 seconds to jump to the following priority... Any ideas to solve this problem? I can't use the IP of the provider (it has a pool of servers), I try to use dnsmgr without solving the isue Why don't you fill the ip addresses to your /etc/hosts file? In that way lookups won't need any dns resolving and still could keep the load balancing by having multiple ip addresses to the same SIP hostname. regards Adam ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anyone else having problems with the list
me too :) Original-Nachricht Datum: Tue, 25 Sep 2007 12:57:25 +0200 Von: Michiel van Baak [EMAIL PROTECTED] An: asterisk-users@lists.digium.com Betreff: Re: [asterisk-users] Anyone else having problems with the list On 22:43, Tue 25 Sep 07, Carlos Hernandez wrote: Yes for me. Carlos -- Julian Lyndon-Smith wrote: I have sent a few emails over the past couple of days that simply have not arrived on the list (or so it seems). Is anyone else encountering this ? Julian I have similar problems. Some mails arrive, some dont. If I check the listarchive on the web I see more emails then in mutt. I already disabled greylisting etc and browsed thru the spam quarantine but nothing. -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called users? ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- GMX FreeMail: 1 GB Postfach, 5 E-Mail-Adressen, 10 Free SMS. Alle Infos und kostenlose Anmeldung: http://www.gmx.net/de/go/freemail ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hola Jonathan, a ver si tre suena...
Am Dienstag, den 25.09.2007, 11:01 + schrieb dadsadsadf dsadasdsa: Hola Jonathan Te cuento un pokillo lo q intento hacer por si me puedes orientar en algo o de algun sitio donde pueda mirar Existe una especificación de Microsoft de lo que llaman Dual-Forking, que básicamente consiste en poder usar tanto el teléfono como el propio PC como dispositivo de comunicaciones, según convenga. De esta manera, por ejemplo, se puede usar estando en la oficina el teléfono de nuestra IP PBX para hacer y recibir llamadas, y sin embargo si se está de viaje, es posible usar el propio PC para iniciar y recibir llamadas a teléfonos IP de una empresa o incluso a RTC a través de su infraestructura de Voz IP. Para gente móvil o en general para perfiles de cliente que se inclinen por una solución softphone, este desarrollo es clave. Para el Dual-Forking, no se necesita tomar algo de Microsoft. Situaciones en que una llamada puede ser recibida como en tu oficina como en softphone de tu ordenador en tu casa si, como en algun telefono real que podria ser situado en todo el mundo - parallel call lo llaman algunos registradores SIP por aqui - es realizado simplementissimo en Asterisk. Por ejemplo, se puede poner en su extensions.conf exten = 201,1,Dial(SIP/officinademartaSIP/martaslaptop,60) y se puede añadir mas telefonos, si SIP si IAX o ZAP - se pone un ampersand entre esos y asterisk prueba llamar todos en mismo momento. Mejor, no hay problema si un de esos no es conectado en ese momento - los otros telefonos van a functionar normalmente. Claro tambien es posible solo llamar al un telefono de que el usador ha puesto la ultima llamada (puede memorar eso en la AstDB, por ejemplo), o miles otras situactiones. Yo tengo un telefono movil que ofrece connexion GSM y WLAN/SIP, que normalmente tomo cuando dejo de casa, y (vale, mas o menos... ;-) dos telefonos fijos conectados a mi asterisk. A vezes (viajando, por ejemplo) tambien tengo un softphone in mi laptop. Tengo que decir que todos son SIP/UDP, pero no puedo imaginar que la software de MS ofrece cosas que no se puede realizar en asterisk. Si es posible para ti, podria ser mejor continuar en ingles - hace algunos años desde aprendio español en el insti secundar (disculpe lo que resulta :-), y la asterisk-users es normalmente usado en ingles, asi puedes recibir mas mensajes de mas gente. Saludo Anselm ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Yikes! Polycom 501 chokes on BootRom 4.0.0?
Doug wrote: I was progressively upgrading this phone from 3.1.2 to 3.2.3, then to 4.0.0. v3.2.3 worked fine, but when I went to 4.0.0 (Even adding the more specific 2345-11500-040.bootrom.ld), it won't run, and just keeps rebooting. Now I've got a really nice doorstop unless someone knows how to get out of this predicament. Help! 0925003705|cfg |3|00|Beginning to provision phone 0925003705|dns |3|00|DNS lookup for 'somedomain.com'(66.16.26.106) TTL=83485 0925003705|copy |3|00|'ftp://someuser:[EMAIL PROTECTED]/2345-11500-040.bootrom.ld' from 'somedomain.com(66.16.26.106)' 0925003706|cfg |3|00|Image 2345-11500-040.bootrom.ld has not changed 0925003706|copy |3|00|Download of '2345-11500-040.bootrom.ld' succeeded on attempt 1 (addr 1 of 1) 0925003706|cfg |3|00|Downloaded bootROM is identical to current version 4.0.0 0925003706|copy |3|00|'ftp://someuser:[EMAIL PROTECTED]/0004f210.cfg' from 'somedomain.com(66.16.26.106)' 0925003707|copy |3|00|Download of '0004f210.cfg' succeeded on attempt 1 (addr 1 of 1) 0925003708|cfg |5|00|Could not get the list of CONFIG_FILES 0925003708|cfg |5|00|Could not get the list of MISC_FILES 0925003709|cfg |5|00|Couldn't get parameter APP_FILE_PATH 0925003709|cfg |3|00|Unspecified error occured with downloaded application image 0925003709|app1 |6|00|Error in saving application. 0925003709|app1 |6|00|Uploading boot log, time is TUE SEP 25 00:37:10 2007 I've upgraded my 501 to bootrom 4.0.0. It did reboot and reformat the filesystem about three times in a row before it finally finished but it did work for me. I'm still using SIP 2.1.2 though. Don't know if that information helps any. -Dave ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Do I need to run #modprobe zaptel for Digium
Hi List; If I am configuring Diguim Analoge card, then I need to run #modprobe wctdm, but the question why I need to run #modprobe zaptel also? What #modprobe zaptel does a things that #modprobe wctdm does not do? Any help? Regards Bilal Looking for a deal? Find great prices on flights and hotels with Yahoo! FareChase. http://farechase.yahoo.com/ ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anyone else having problems with the list
On Tue, 2007-09-25 at 10:14 +0100, Julian Lyndon-Smith wrote: I have sent a few emails over the past couple of days that simply have not arrived on the list (or so it seems). I'll take a look at this again... I thought we had most of the problems with the mailing lists fixed, but we seem to be having some problems again. (This is most likely our spam-catching system being over-aggressive, but I'll look into it.) -- Jared Smith Community Relations Manager Digium, Inc. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help with Sip Registration
On Wed, 2007-09-26 at 03:49 +0630, Treesa Fairy Joseph wrote: Hi all, I have installed X-lite client on a windowsXP machine and asterisk on an enterprise linux m/c. The client is sending a registration message to asterisk server. It is able to process the message and sends 200 OK back. This is working correctly, and your X-lite client is successfully registered to Asterisk. But later it says Scheduling destruction of sip dialog . Then it says Really destroying sip dialog . What to do for this problem??? This is not an error. This is simple an informative message from the SIP channel driver, and you can safely ignore it. -- Jared Smith Community Relations Manager Digium, Inc. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Do I need to run #modprobe zaptel for Digium
On Tue, Sep 25, 2007 at 05:55:13AM -0700, bilal ghayyad wrote: Hi List; If I am configuring Diguim Analoge card, then I need to run #modprobe wctdm, but the question why I need to run #modprobe zaptel also? No. What #modprobe zaptel does a things that #modprobe wctdm does not do? modprobe will load all the modules on which your module depends first. wctdm depends on zaptel, and hence it would first load zaptel and later load wctdm. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Do I need to run #modprobe zaptel for Digium
On Tue, 2007-09-25 at 05:55 -0700, bilal ghayyad wrote: If I am configuring Diguim Analoge card, then I need to run #modprobe wctdm, but the question why I need to run #modprobe zaptel also? The wctdm kernel module depends on the zaptel module, so the zaptel module will get automatically loaded when you load the wctdm module. -- Jared Smith Community Relations Manager Digium, Inc. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF dropping digits
In article [EMAIL PROTECTED], Barton Fisher [EMAIL PROTECTED] wrote: We have a Te410P with 3 Telco T1's (D4 SF ) with DID's (non-PRI). ANI DNIS is received in-band DTMF in a format such as *7145551212*8002* What happens when there are 30 or more calls, asterisk might see is DNIS = 802 or ANI = 4551212 for examples, where parts of the numbers are dropped. All the traffic arrives into a simple IVR script where a message is played. We are using Asterisk 1.2 and Server is 2.8 Dual Xeon SuperMicro with 2 GB RAM. Any clues what I can do to fix this? Try applying the patch at http://bugs.digium.com/view.php?id=10535 Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Redundancy
Adrian Marsh wrote: I'm interested in how people are clustering Asterisk, if that's possible, or how you might be achieving a redundant solution. I've a single Asterisk server driving the company. Its well backed-up, and I've a cloned machine that (in theory) with a DNS change could take over operations. However I'd like to achieve something more automated if possible. I haven't looked into it in any detail, but how about the standard Linux HA solution with a heartbeat monitor, a shared file-system and IP take-over? /Per Jessen, Zürich -- http://www.spamchek.com/ - your spam is our business. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Zaptel-1.4.5.1 Compile Error
Hi All, I'm compiling zaptel. Did the usual ./configure, then make. Compile breaks saying: /usr/src/zaptel-1.4.5.1/wcusb.c:1451: error: unknown field âownerâ specified in initializer /usr/src/zaptel-1.4.5.1/wcusb.c:1451: warning: initialization from incompatible pointer type make[3]: *** [/usr/src/zaptel-1.4.5.1/wcusb.o] Error 1 make[2]: *** [_module_/usr/src/zaptel-1.4.5.1] Error 2 What am I missing here? Do I have to have my digium card installed first before compiling zaptel? I am running Fedora Core 5. Thanks for your help. Jeng ___ Want ideas for reducing your carbon footprint? Visit Yahoo! For Good http://uk.promotions.yahoo.com/forgood/environment.html ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ExternNotify Voicemail
Nevermind, I found the answer on the wiki: Want to run an external program whenever a caller leaves a voice mail message for a user? This is where the externnotify command comes in handy. Externnotify takes a string value which is the command line you want to execute when the caller finishes leaving a message. Note: see an example of an external notification script here. Note: This command will also run after a person who has logged into a mailbox exits the VoiceMailMain() application. The way it works is basically any time that somebody leaves a voicemail on the system (regardless of mailbox number), the command specified for externnotify is run with the arguments (in this order): context, extension, and number of voicemails in that mailbox. These arguments are passed to the program that you set in the externnotify variable. But, it would be nice to have one of the arguments be what event triggered the script. Like if it was a message was left, or some logged out of VoicemailAdmin Forrest Beck [EMAIL PROTECTED] www.shift8.biz On Sep 24, 2007, at 10:36 PM, Forrest Beck wrote: I have googled and can seem to find the answer to this one Does anyone here have experience with externnotify in voicemail.conf? The sample states that it will run when a message is delivered and retrieved. Does asterisk pass any arguments to the script? Thanks. Forrest Beck [EMAIL PROTECTED] www.shift8.biz ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Redundancy
On Tue, 2007-09-25 at 15:59 +0200, Per Jessen wrote: I haven't looked into it in any detail, but how about the standard Linux HA solution with a heartbeat monitor, a shared file-system and IP take-over? It's been my experience that this usually works fairly well for stateless protocols like HTTP, but doesn't do so well on stateful protocols like SIP and IAX, and in general is a much more difficult problem to solve. Most people tend to use some combination of SIP proxies (such as SER and OpenSER), DUNDi, shared storage, redundant databases with replication, T1/E1 failover boxes, and horizontal scaling to make Asterisk more highly-available. Of course, I haven't really gone into much detail here, but hopefully it helps answer your question. (It's also my personal experience that people who know how to build such solutions are making enough money off of selling their solution that they aren't real eager to give away all their secrets.) In reality though, you say the word cluster and it means five different things to five different people. To really be able to answer the original poster's question, we'd really have to know a lot more about his architecture and his potential points of failure. -- Jared Smith Community Relations Manager Digium, Inc. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] show queue (queue name)
Hi all, does anybody know any way that when it run reload app_queue in the asterisk cli it don't lose the informations from show queue (queue name) ? I'm passing for this trouble, because I need this informations (http://www.voip-info.org/wiki/index.php?page=asterisk+cli+command+show+queue) that asterisk cli command show queue (queue name) show me for my external application, but when I need to include a new queue or agent I need to restart the app_queue, than the asterisk lost this informations and begin with an empty set. there is a way to resolve this? if anybody knows please give me this informations or hints to revolse this... thank's a lot for the opportunity Everton Goularth GOVoIP www.govoip.com.br ___ Yahoo! Mail - Sempre a melhor opção para você! Experimente já e veja as novidades. http://br.yahoo.com/mailbeta/tudonovo/ ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Multiple Home system with SIP
Is there a way to tell asterisk, via a sip.conf peer, what IP address to send a packet out of? I've got multiple NICs in my box, each with it's own public IP. I need the SIP messages to originate from any one of the IPs depending on which number was originally called(and therefore where the packet originally came from). My fear is that it will listen on all IPs fine, but only respond via the default GW. This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel-1.4.5.1 Compile Error
On Tue, Sep 25, 2007 at 03:22:01PM +0100, Jeng Yu wrote: Hi All, I'm compiling zaptel. Did the usual ./configure, then make. Compile breaks saying: /usr/src/zaptel-1.4.5.1/wcusb.c:1451: error: unknown field âownerâ specified in initializer /usr/src/zaptel-1.4.5.1/wcusb.c:1451: warning: initialization from incompatible pointer type make[3]: *** [/usr/src/zaptel-1.4.5.1/wcusb.o] Error 1 make[2]: *** [_module_/usr/src/zaptel-1.4.5.1] Error 2 What am I missing here? Do I have to have my digium card installed first before compiling zaptel? I am running Fedora Core 5. What kernel version? uname -r -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] swift.conf - cepstral voice quality adjustment options
Hi all,
I hope that I'm not breaking protocol too much by posting a message in this
group about a problem that I'm having with an Asterisk Business Edition
installation, but the reason that I'm posting here is because the problem
that I'm having isn't really with the Business Edition, it is with the
Cepstral text to speech product that I'm using with it, and also because this
group has so much more activity that I'm really hoping that somewhere in
this great Asterisk community there are some clever people who might have
some good suggestions to help me improve the voice quality on this system.
I have a phone switch configured with Asterisk Business Edition (B.2.3-2-1)
and I have installed a single license for the Cepstral voice, Callie. I am
using the Swift application call within the asterisk dialplan:
Within /etc/asterisk/extensions.conf --
exten = s,n,Swift(Callie^Hello this is a cepstral voice named
Callie.)
And the php class phpAGI2, for interfacing with the Asterisk Gateway
Interface:
Within /var/lib/asterisk/agi-bin/lookupAgency.php --
$agi-swift('The agency nearest to your store is '.$agency_name);
Both of these calls use the Callie voice and translate the text into speech,
but the voice quality seems to be less than ideal. The voice sounds much
more 'robotic' than the demonstration on the Cepsral site, and the
pronunciation is noticeably slurred in areas. I have looked and looked all
over to try and find documentation on how I can make any improvements to the
voice quality. I have contacted Digium and Cepstral, but neither had much
to offer in the way of support. I have googled and googled, but I can't
seem to find anything that helps. I see that there is a file called
'/etc/asterisk/swift.conf' but all that is in it is the single entry:
[general]
voice=Callie; the name of the Cepstral voice
I'm wondering if there are other swift.conf entries that I might be able to
use to help improve the voice quality? Command line options?? Or, maybe
there are other options available?? I am open to anything that might help.
Anyone who is interested in hearing the quality can call in and hear it at
865-288-6300. This is an IVR system that will be rolled out to support
World Hunger Week. Stores will begin calling in during the week of October
1st, so we are very near the roll out date.
I have not yet purchased the concurrency license from Cepstral because of
the voice quality concerns. So if more than one call is received
simultaneously right now, all but the first will hear a licensing alert
along with the scripted text.
Any help or suggestions as to what I can do to improve the voice quality is
greatly appreciated.
Sincerely,
Larry Costigan
Food Donation Connection
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Re: [asterisk-users] Asterisk Redundancy
Sure, Heres a basic overview: - All IP (no local E1/T1 connections). - 2Mb Fiber internet pipe backed up by a DSL backup. - Single Asterisk server (with a backup clone on standby). Config currently backed up to SVN and copied off by tarball by webmin to a separate network. - Both IAX and SIP connectivity to 2 providers, with A*k Dial command driven failover for outbound calls (PSTN inbound limited to one provider). - All UPS backed. That's about the current config. This is an office/company config, not a reseller. Main points of failure: Fiber/DSL Box (easy to swap out). Same for the Fiber/DSL lines themselves. The main A*k box itself. I've covered all the redundancy I can (within budget) of the connectivity, and I'm wondering what I can do with A*k itself. I'm guessing that SIP proxies might be overkill (as I'd then need redundancy within those too), so maybe it's a case of looking at Linux-HA. Adrian Marsh -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jared Smith Sent: 25 September 2007 15:28 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk Redundancy On Tue, 2007-09-25 at 15:59 +0200, Per Jessen wrote: I haven't looked into it in any detail, but how about the standard Linux HA solution with a heartbeat monitor, a shared file-system and IP take-over? It's been my experience that this usually works fairly well for stateless protocols like HTTP, but doesn't do so well on stateful protocols like SIP and IAX, and in general is a much more difficult problem to solve. Most people tend to use some combination of SIP proxies (such as SER and OpenSER), DUNDi, shared storage, redundant databases with replication, T1/E1 failover boxes, and horizontal scaling to make Asterisk more highly-available. Of course, I haven't really gone into much detail here, but hopefully it helps answer your question. (It's also my personal experience that people who know how to build such solutions are making enough money off of selling their solution that they aren't real eager to give away all their secrets.) In reality though, you say the word cluster and it means five different things to five different people. To really be able to answer the original poster's question, we'd really have to know a lot more about his architecture and his potential points of failure. -- Jared Smith Community Relations Manager Digium, Inc. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Redundancy
Adrian Marsh wrote: I'm interested in how people are clustering Asterisk, if that's possible, or how you might be achieving a redundant solution. I've a single Asterisk server driving the company. Its well backed-up, and I've a cloned machine that (in theory) with a DNS change could take over operations. However I'd like to achieve something more automated if possible. Maybe my post at http://lists.digium.com/pipermail/asterisk-users/2007-August/195339.html could provide you with some answers. I don't want to quote my text as not to spam the list (although it's all GPL). There's a nice countdown at http://www.amooma.de/gemeinschaft/ but we're all quite busy at the moment (that page still needs to be translated). Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de My pick of the month: rfc 2822 3.6.5 Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] swift.conf - cepstral voice quality adjustment options
On Tue, 2007-09-25 at 10:57 -0400, Larry Costigan wrote: Hi all, I hope that I'm not breaking protocol too much by posting a message in this group about a problem that I'm having with an Asterisk Business Edition installation, but the reason that I'm posting here is because the problem that I'm having isn't really with the Business Edition, it is with the Cepstral text to speech product that I'm using with it, and also because this group has so much more activity that I'm really hoping that somewhere in this great Asterisk community there are some clever people who might have some good suggestions to help me improve the voice quality on this system. SNIP Here is a link that provides a snippet of info for you. http://www.mezzo.net/asterisk/app_swift.html I would think that the buffer setting might be of importance! dave ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel-1.4.5.1 Compile Error
Sorry, I should have mentioned it in my mail. uname -r gives: 2.6.15-1.2054_FC5smp Thanks, Jeng --- Tzafrir Cohen [EMAIL PROTECTED] wrote: On Tue, Sep 25, 2007 at 03:22:01PM +0100, Jeng Yu wrote: Hi All, I'm compiling zaptel. Did the usual ./configure, then make. Compile breaks saying: /usr/src/zaptel-1.4.5.1/wcusb.c:1451: error: unknown field âownerâ specified in initializer /usr/src/zaptel-1.4.5.1/wcusb.c:1451: warning: initialization from incompatible pointer type make[3]: *** [/usr/src/zaptel-1.4.5.1/wcusb.o] Error 1 make[2]: *** [_module_/usr/src/zaptel-1.4.5.1] Error 2 What am I missing here? Do I have to have my digium card installed first before compiling zaptel? I am running Fedora Core 5. What kernel version? uname -r -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Yahoo! Answers - Got a question? Someone out there knows the answer. Try it now. http://uk.answers.yahoo.com/ ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] show queue (queue name)
Everton Goularth wrote: does anybody know any way that when it run reload app_queue in the asterisk cli it don't lose the informations from show queue (queue name) ? I'm passing for this trouble, because I need this informations (http://www.voip-info.org/wiki/index.php?page=asterisk+cli+command+show+queue) that asterisk cli command show queue (queue name) show me for my external application, but when I need to include a new queue or agent I need to restart the app_queue, than the asterisk lost this informations and begin with an empty set. there is a way to resolve this? if anybody knows please give me this informations or hints to revolse this... Realtime. http://www.voip-info.org/wiki-Asterisk+RealTime There's no need to reload app_queue when using Realtime. Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de My pick of the month: rfc 2822 3.6.5 Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel-1.4.5.1 Compile Error
Also, /usr/bin/gcc --version gives: gcc (GCC) 4.1.0 20060304 (Red Hat 4.1.0-3) Copyright (C) 2006 Free Software Foundation, Inc. Also, /usr/bin/make --version gives: GNU Make 3.80 Copyright (C) 2002 Free Software Foundation, Inc. Thanks, Jeng --- Jeng Yu [EMAIL PROTECTED] wrote: Hi All, I'm compiling zaptel. Did the usual ./configure, then make. Compile breaks saying: /usr/src/zaptel-1.4.5.1/wcusb.c:1451: error: unknown field âownerâ specified in initializer /usr/src/zaptel-1.4.5.1/wcusb.c:1451: warning: initialization from incompatible pointer type make[3]: *** [/usr/src/zaptel-1.4.5.1/wcusb.o] Error 1 make[2]: *** [_module_/usr/src/zaptel-1.4.5.1] Error 2 What am I missing here? Do I have to have my digium card installed first before compiling zaptel? I am running Fedora Core 5. Thanks for your help. Jeng ___ Want ideas for reducing your carbon footprint? Visit Yahoo! For Good http://uk.promotions.yahoo.com/forgood/environment.html ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Want ideas for reducing your carbon footprint? Visit Yahoo! For Good http://uk.promotions.yahoo.com/forgood/environment.html ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ISDN2#02: too much voice to send for NCCI=0x40502
My problem: Sometimes the sound seems to cut on calls in progress. We (on our local SIP phones, Thomson ST2030's) can't hear the remote caller. The caller may hear some kind of horrid sklurk and then it goes dead for them too. Our Asterisk is connected to the France Telecom network by an Eicon Diva Server 4xBRI (we have 3 BRI lines). Here's what I've found by turning on huge amounts of debug stuff: everything runs just tickety-boo until the remote end seems to stop sending us packets, our packets seem not to be sent and we get the infamous too much voice to send error. Help! What might be going on? Here's an example: [ All is well: ] Got RTP packet from192.168.6.185:41000 (type 08, seq 018498, ts 1479920, len 80) Got RTP packet from192.168.6.185:41000 (type 08, seq 018499, ts 148, len 80) DATA_B3_REQ ID=002 #0xee37 LEN=0030 Controller/PLCI/NCCI= 0x40502 Data32 = 0x0 DataLength = 0xa0 DataHandle = 0xb624 Flags = 0x0 Data64 = 0x82fa70 DATA_B3_CONF ID=002 #0xee37 LEN=0016 Controller/PLCI/NCCI= 0x40502 DataHandle = 0xb624 Info= 0x0 DATA_B3_IND ID=002 #0x8a51 LEN=0030 Controller/PLCI/NCCI= 0x40502 Data32 = 0x0 DataLength = 0xa0 DataHandle = 0x49 Flags = 0x0 Data64 = 0x2aaac7caa49e DATA_B3_RESP ID=002 #0x8a51 LEN=0014 Controller/PLCI/NCCI= 0x40502 DataHandle = 0x49 -- ISDN2#02: DATA_B3_IND (len=160) fr.datalen=160 fr.subclass=8 Sent RTP packet to 192.168.6.185:41000 (type 08, seq 064338, ts 1428080, len 000160) [ but at this point we seem to stop getting data from the network. We continue to send: ] Got RTP packet from192.168.6.185:41000 (type 08, seq 018500, ts 1480080, len 80) Got RTP packet from192.168.6.185:41000 (type 08, seq 018501, ts 1480160, len 80) DATA_B3_REQ ID=002 #0xee39 LEN=0030 Controller/PLCI/NCCI= 0x40502 Data32 = 0x0 DataLength = 0xa0 DataHandle = 0xb625 Flags = 0x0 Data64 = 0x82fb50 DATA_B3_CONF ID=002 #0xee39 LEN=0016 Controller/PLCI/NCCI= 0x40502 DataHandle = 0xb625 Info= 0x0 [... 8 RTP packets, 4 DATA_B3_REQ's omitted ] [ And then: ] Got RTP packet from192.168.6.185:41000 (type 08, seq 018510, ts 1480880, len 80) Got RTP packet from192.168.6.185:41000 (type 08, seq 018511, ts 1480960, len 80) ISDN2#02: too much voice to send for NCCI=0x40502 Got RTP packet from192.168.6.185:41000 (type 08, seq 018512, ts 1481040, len 80) Got RTP packet from192.168.6.185:41000 (type 08, seq 018513, ts 1481120, len 80) ISDN2#02: too much voice to send for NCCI=0x40502 [... Someone gets fed up and cuts the B channel: ] Got RTP packet from192.168.6.185:41000 (type 08, seq 018610, ts 140, len 80) Got RTP packet from192.168.6.185:41000 (type 08, seq 018611, ts 1488960, len 80) ISDN2#02: too much voice to send for NCCI=0x40502 DISCONNECT_B3_IND ID=002 #0x8a8a LEN=0015 Controller/PLCI/NCCI= 0x40502 Reason_B3 = 0x0 NCPI= default DISCONNECT_B3_RESP ID=002 #0x8a8a LEN=0012 Controller/PLCI/NCCI= 0x40502 End of conversation. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel-1.4.5.1 Compile Error
Upgrade your kernel. Run: # uname -r if you do not see smp in the kernel version Run: # yum update kernel kernel-devel If you do see smp Run: # yum update kernel-smp kernel-smp-devel Forrest Beck [EMAIL PROTECTED] www.shift8.biz On Sep 25, 2007, at 10:53 AM, Tzafrir Cohen wrote: On Tue, Sep 25, 2007 at 03:22:01PM +0100, Jeng Yu wrote: Hi All, I'm compiling zaptel. Did the usual ./configure, then make. Compile breaks saying: /usr/src/zaptel-1.4.5.1/wcusb.c:1451: error: unknown field âownerâ specified in initializer /usr/src/zaptel-1.4.5.1/wcusb.c:1451: warning: initialization from incompatible pointer type make[3]: *** [/usr/src/zaptel-1.4.5.1/wcusb.o] Error 1 make[2]: *** [_module_/usr/src/zaptel-1.4.5.1] Error 2 What am I missing here? Do I have to have my digium card installed first before compiling zaptel? I am running Fedora Core 5. What kernel version? uname -r -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ Sign up now for AstriCon 2007! September 25-28th. http:// www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Redundancy
Adrian Marsh wrote: so maybe it's a case of looking at Linux-HA. If I remember this correctly a normal ping is all Linux HA can do. It does not check whether Asterisk or other services are alive and respond to queries. Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de My pick of the month: rfc 2822 3.6.5 Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] show queue (queue name)
On 9/25/07, Everton Goularth [EMAIL PROTECTED] wrote: does anybody know any way that when it run reload app_queue in the asterisk cli it don't lose the informations from show queue (queue name) ? A 'keepstats' option has been added to -trunk, and will show up when 1.6 is released. Until then, you'd have to look at backporting this change: http://svn.digium.com/view/asterisk/trunk/apps/app_queue.c?r1=43945r2=44150 (it's a pretty small change) -- j. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Redundancy
On Tue, 2007-09-25 at 18:01 +0200, Philipp Kempgen wrote: Adrian Marsh wrote: so maybe it's a case of looking at Linux-HA. If I remember this correctly a normal ping is all Linux HA can do. It does not check whether Asterisk or other services are alive and respond to queries. Have you looked at: http://www.voip-info.org/wiki-Asterisk+monitoring My personal favourite would be nagios (not that I have used the SIP plugin, but do use nagios for other services) Kind Regards, Dave Walker signature.asc Description: This is a digitally signed message part ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel-1.4.5.1 Compile Error
On Tuesday 25 September 2007 09:22:01 Jeng Yu wrote: /usr/src/zaptel-1.4.5.1/wcusb.c:1451: error: unknown field âownerâ specified in initializer Type 'make menuselect', deselect wcusb, then left-arrow out to the top, hit 's' for save, then 'make' again. -- Tilghman ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Redundancy
Dave Walker wrote: On Tue, 2007-09-25 at 18:01 +0200, Philipp Kempgen wrote: Adrian Marsh wrote: so maybe it's a case of looking at Linux-HA. If I remember this correctly a normal ping is all Linux HA can do. It does not check whether Asterisk or other services are alive and respond to queries. Have you looked at: http://www.voip-info.org/wiki-Asterisk+monitoring My personal favourite would be nagios (not that I have used the SIP plugin, but do use nagios for other services) Exactly. If this was about monitoring I'd suggest to have a look at Nagios. But it's quite easy to write your own script which checks if Asterisk responds to SIP packets (or whatever) and takes over the IP address of your main server once Asterisk fails to reply. Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de My pick of the month: rfc 2822 3.6.5 Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Redundancy
It's nice to see Asterisk redundancy being discussed. A year and half ago, when I posed the question of Asterisk redundancy, I was looked at like I was from outer space. - Original Message From: Jared Smith [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, September 25, 2007 7:27:37 AM Subject: Re: [asterisk-users] Asterisk Redundancy On Tue, 2007-09-25 at 15:59 +0200, Per Jessen wrote: I haven't looked into it in any detail, but how about the standard Linux HA solution with a heartbeat monitor, a shared file-system and IP take-over? It's been my experience that this usually works fairly well for stateless protocols like HTTP, but doesn't do so well on stateful protocols like SIP and IAX, and in general is a much more difficult problem to solve. Most people tend to use some combination of SIP proxies (such as SER and OpenSER), DUNDi, shared storage, redundant databases with replication, T1/E1 failover boxes, and horizontal scaling to make Asterisk more highly-available. Of course, I haven't really gone into much detail here, but hopefully it helps answer your question. (It's also my personal experience that people who know how to build such solutions are making enough money off of selling their solution that they aren't real eager to give away all their secrets.) In reality though, you say the word cluster and it means five different things to five different people. To really be able to answer the original poster's question, we'd really have to know a lot more about his architecture and his potential points of failure. -- Jared Smith Community Relations Manager Digium, Inc. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Check out the hottest 2008 models today at Yahoo! Autos. http://autos.yahoo.com/new_cars.html___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Grandstream GXP2020 / 2000
On Tue, 2007-09-25 at 10:59 +0200, Erik Wartusch wrote: Hi, Has somebody experiences with the Grandstream GXP2020 / 2000 phones in a business graded installation (with really traffic on not 3 calls a day ;-) ) Of course with Asterisk PBX (1.4.1 or 1.4.11 or 1.4 in generall) I have not tried the 2020 yet but the GXP-2000 works fairly well. The only complaint I had from a very busy installation (a travel agency) is that the handset gets hot after prolonged use. This may have been because the office itself was hot during summer and after they installed an AC the problem is no longer there. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Redundancy
Nagios that's not redundancy. - Original Message From: Dave Walker [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, September 25, 2007 9:09:46 AM Subject: Re: [asterisk-users] Asterisk Redundancy On Tue, 2007-09-25 at 18:01 +0200, Philipp Kempgen wrote: Adrian Marsh wrote: so maybe it's a case of looking at Linux-HA. If I remember this correctly a normal ping is all Linux HA can do. It does not check whether Asterisk or other services are alive and respond to queries. Have you looked at: http://www.voip-info.org/wiki-Asterisk+monitoring My personal favourite would be nagios (not that I have used the SIP plugin, but do use nagios for other services) Kind Regards, Dave Walker Tonight's top picks. What will you watch tonight? Preview the hottest shows on Yahoo! TV. http://tv.yahoo.com/ ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Grandstream GXP2020 / 2000
I have a client using the Grandstream phones (not sure which model but it looks fairly low-end) and they're lukewarm on them. The display doesn't tilt up for easy viewing and the sound quality on the speaker phone leaves something to be desired apparently. But as basic, inexpensive, Asterisk handsets they do the job. Ben M. Schorr Chief Executive Officer __ Roland Schorr Tower www.rolandschorr.com [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Carlos Chavez Sent: Tuesday, September 25, 2007 7:08 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Grandstream GXP2020 / 2000 On Tue, 2007-09-25 at 10:59 +0200, Erik Wartusch wrote: Hi, Has somebody experiences with the Grandstream GXP2020 / 2000 phones in a business graded installation (with really traffic on not 3 calls a day ;-) ) Of course with Asterisk PBX (1.4.1 or 1.4.11 or 1.4 in generall) I have not tried the 2020 yet but the GXP-2000 works fairly well. The only complaint I had from a very busy installation (a travel agency) is that the handset gets hot after prolonged use. This may have been because the office itself was hot during summer and after they installed an AC the problem is no longer there. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Redundancy
On Tue, 2007-09-25 at 12:10 -0500, Douglas Garstang wrote: Nagios that's not redundancy. And a brick isn't a house. Clearly you know what Nagios is; and it's support for event-handlers. If you had taken a moment to think, then you would know Nagios can form part of a redundancy system. signature.asc Description: This is a digitally signed message part ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Yikes! Polycom 501 chokes on BootRom 4.0.0?
I am having a similar issue with 4.0.0. Mine is that it doesn't get any DHCP address (gets stuck waiting for an address). I fixed it by going back one to the previous bootrom version, worked like a charm. Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dave Fullerton Sent: Tuesday, September 25, 2007 08:49 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Yikes! Polycom 501 chokes on BootRom 4.0.0? Doug wrote: I was progressively upgrading this phone from 3.1.2 to 3.2.3, then to 4.0.0. v3.2.3 worked fine, but when I went to 4.0.0 (Even adding the more specific 2345-11500-040.bootrom.ld), it won't run, and just keeps rebooting. Now I've got a really nice doorstop unless someone knows how to get out of this predicament. Help! 0925003705|cfg |3|00|Beginning to provision phone dns |3|00|DNS 0925003705|lookup for 'somedomain.com'(66.16.26.106) TTL=83485 copy |3|00|'ftp://someuser:[EMAIL PROTECTED]/2345-11500-040.bootrom.ld' from 'somedomain.com(66.16.26.106)' 0925003706|cfg |3|00|Image 2345-11500-040.bootrom.ld has not changed 0925003706|copy |3|00|Download of '2345-11500-040.bootrom.ld' succeeded on attempt 1 (addr 1 of 1) 0925003706|cfg |3|00|Downloaded bootROM is identical to current 0925003706|version 4.0.0 copy |3|00|'ftp://someuser:[EMAIL PROTECTED]/0004f210.cfg' from 'somedomain.com(66.16.26.106)' 0925003707|copy |3|00|Download of '0004f210.cfg' succeeded on attempt 1 (addr 1 of 1) 0925003708|cfg |5|00|Could not get the list of CONFIG_FILES cfg 0925003708||5|00|Could not get the list of MISC_FILES 0925003709|cfg |5|00|Couldn't get parameter APP_FILE_PATH cfg 0925003709||3|00|Unspecified error occured with downloaded application image 0925003709|app1 |6|00|Error in saving application. 0925003709|app1 |6|00|Uploading boot log, time is TUE SEP 25 00:37:10 0925003709|2007 I've upgraded my 501 to bootrom 4.0.0. It did reboot and reformat the filesystem about three times in a row before it finally finished but it did work for me. I'm still using SIP 2.1.2 though. Don't know if that information helps any. -Dave ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Redundancy
Philipp Kempgen wrote: Adrian Marsh wrote: so maybe it's a case of looking at Linux-HA. If I remember this correctly a normal ping is all Linux HA can do. It does not check whether Asterisk or other services are alive and respond to queries. I think the basic Linux-HA setup works with ping, but there's plenty of applications (mysql, apache, mailservers) that have their own plugins to monitor application level availability. /Per Jessen, Zürich -- http://www.spamchek.com/ - your spam is our business. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Redundancy
Philipp Kempgen wrote: I don't want to quote my text as not to spam the list (although it's all GPL). There's a nice countdown at http://www.amooma.de/gemeinschaft/ Very nice. I'll have to come back and take a closer look sometime. /Per Jessen, Zürich -- http://www.spamchek.com/ - your spam is our business. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Grandstream GXP2020 / 2000
On Tue, 25 Sep 2007, Ben Schorr wrote: I have a client using the Grandstream phones (not sure which model but it looks fairly low-end) and they're lukewarm on them. The display doesn't tilt up for easy viewing and the sound quality on the speaker phone leaves something to be desired apparently. Sounds like the BudgeTone 100/101/200 phones. They are numeric only displays which don't tilt. The GXP2000's do tilt. The early versions of the BT100/101/200 firmware had problems and I could make them hang requiring a power cycle - but the latest s/w (1.0.8.33) seems to have fixed this. (And until very recently I'd never have delpoyed a BT200 to a customer because of this!) But as basic, inexpensive, Asterisk handsets they do the job. I've deployed quite a few GXP2000's this year. Biggest single installation is only 10, but I have one in 2 weeks time where they want 20 in the office. I'm not anticipating any problems. They may not be the best, but for the price are more than adequate. Problems I have heard of (but not experienced - yet?) are handsets getting warm - I think this was an early hardware fault though. Backlights getting feint, and low screen contrast (the display is hard to read when the backlight is off - it can be turned on permanently in via the web interface) The speakerphone on the 2000 is adequate, but the microphone gain is probably set too high (complaints of too much background noise from the other party) Early 2000's had issues too, but the current s/w (1.1.1.14) has been very stable, although the BLF functions on the GXP2000 mostly work in the current version I have, sometime gets confused. The phone itself picks up mains hum when my desk lamp is right next to it (but so does the Snom!) Features that the users like is the easy way to shuttle between 2 calls with the line buttons at the top of the phone. (I don't use the multi-account features though - the only one I've installed with more than one account is the one on my desk!) My busyest site with GXP's has one person taking about 20-30 calls a day with one. Hope this helps, Gordon Ben M. Schorr Chief Executive Officer __ Roland Schorr Tower www.rolandschorr.com [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Carlos Chavez Sent: Tuesday, September 25, 2007 7:08 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Grandstream GXP2020 / 2000 On Tue, 2007-09-25 at 10:59 +0200, Erik Wartusch wrote: Hi, Has somebody experiences with the Grandstream GXP2020 / 2000 phones in a business graded installation (with really traffic on not 3 calls a day ;-) ) Of course with Asterisk PBX (1.4.1 or 1.4.11 or 1.4 in generall) I have not tried the 2020 yet but the GXP-2000 works fairly well. The only complaint I had from a very busy installation (a travel agency) is that the handset gets hot after prolonged use. This may have been because the office itself was hot during summer and after they installed an AC the problem is no longer there. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Point-to-Point SIP link without registration
Greetings list, I need to set up a point to point SIP connection between two devices without either of them registering with a registrar/proxy/etc. at all. The devices I've tested so far all seem to insist on having a registration before they'll make or take calls. One of the devices needs to be an ATA with an FXO port (e.g. Sipura/Linksys SPA-3000/3102), the other device can be either an ATA or a SIP Phone. Does anyone have any hardware recommendations that'll work in this scenario? Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited For full contact details visit http://www.minotaur.it This email is made from 100% recycled electrons ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Point-to-Point SIP link without registration
Run it with a stock OpenSER installation that will accept registrations and acknowledge them with a 200 OK, but not actually do anything with them. On Tue, 25 Sep 2007, Chris Bagnall wrote: Greetings list, I need to set up a point to point SIP connection between two devices without either of them registering with a registrar/proxy/etc. at all. The devices I've tested so far all seem to insist on having a registration before they'll make or take calls. One of the devices needs to be an ATA with an FXO port (e.g. Sipura/Linksys SPA-3000/3102), the other device can be either an ATA or a SIP Phone. Does anyone have any hardware recommendations that'll work in this scenario? Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited For full contact details visit http://www.minotaur.it This email is made from 100% recycled electrons ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multiple Home system with SIP
JM == Jeremy Mann [EMAIL PROTECTED] writes: I would have answered, but I was prohibited from quoting properly: JM If you are the intended recipient, further disclosures are JM prohibited without proper authorization. /Benny ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multiple Home system with SIP
Why did you waste time with this reply? You do realize some users don't have control over their Exchange servers, and asinine footers are placed into an email without their intervention or control right? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Benny Amorsen Sent: Tuesday, September 25, 2007 1:55 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Multiple Home system with SIP JM == Jeremy Mann [EMAIL PROTECTED] writes: I would have answered, but I was prohibited from quoting properly: JM If you are the intended recipient, further disclosures are JM prohibited without proper authorization. /Benny This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multiple Home system with SIP
since asterisk is only using operating system's routing ability , you can always set static routes using route command in linux . On 26/09/2007, Jeremy Mann [EMAIL PROTECTED] wrote: Why did you waste time with this reply? You do realize some users don't have control over their Exchange servers, and asinine footers are placed into an email without their intervention or control right? -Original Message- From: [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] On Behalf Of Benny Amorsen Sent: Tuesday, September 25, 2007 1:55 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Multiple Home system with SIP JM == Jeremy Mann [EMAIL PROTECTED] writes: I would have answered, but I was prohibited from quoting properly: JM If you are the intended recipient, further disclosures are JM prohibited without proper authorization. /Benny This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Do I need to run #modprobe zaptel for Digium
Bilal, The '#' symbol is part of a root prompt, not the command. In fact, if you run a command in this way, it will not work because the shell will perceive you as trying to enter a comment, as one would do in a shellscript. The Zapata modules have a series of interdependencies based on the logical decomposition of the code; it is not necessary that the zaptel module do anything from a functional point of view, as far as interacting with the hardware, for it to be a necessary dependency. It may implement certain global utility or API functions -- symbols the wctdm module needs to do some of its work and interface with the rest of the Zapata layer. Programming is an abstract thing. :-) Take care, -- Alex On Tue, 25 Sep 2007, bilal ghayyad wrote: Hi List; If I am configuring Diguim Analoge card, then I need to run #modprobe wctdm, but the question why I need to run #modprobe zaptel also? What #modprobe zaptel does a things that #modprobe wctdm does not do? Any help? Regards Bilal Looking for a deal? Find great prices on flights and hotels with Yahoo! FareChase. http://farechase.yahoo.com/ ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multiple Home system with SIP
And if the Sip provider is sending data from 1 or two fixed hosts? For instance, they send DID1 to IP A.B.C.D from 1.1.1.1 They send DID2 to IP E.F.G.H from 1.1.1.1 How do you differentiate? Would fromhost= work? This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dutch Number for Inbound
A friend of mine just sent me this email - he is looking for an IAX inbound service in Holland - any thoughts? Voip info only has Nadiz which seems to be SIP only. Hi Dean, I need a Dutch number with IAX support. Do you have any leads in that direction? It's been difficult for me to figure it out -- especially since most of their sites seem to be in Dutch... Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] +1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial). ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] HOWTO/FAQ question (Location: Sweden)
zoachien == zoachien [EMAIL PROTECTED] writes: zoachien Turbo Fredriksson wrote: How do I connect to a 'normal' (i.e. analog) telephone? zoachien - you can take a voip provider and not buy any hardware. I was thinking in this way to, but I was unsure if I can still use Asterisk in all it's glory (i.e. with all the cool modules like MP3 player, call center stuff etc), or will this be in the hands of the telco? ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Redundancy
On 9/25/07, Philipp Kempgen [EMAIL PROTECTED] wrote: Adrian Marsh wrote: I'm interested in how people are clustering Asterisk, if that's possible, or how you might be achieving a redundant solution. I've a single Asterisk server driving the company. Its well backed-up, and I've a cloned machine that (in theory) with a DNS change could take over operations. However I'd like to achieve something more automated if possible. Maybe my post at http://lists.digium.com/pipermail/asterisk-users/2007-August/195339.html could provide you with some answers. Hi, This seems nice way of sharing settings, however it wouldn't take over calls in progress. For us, currently the greatest problem is that whenever Asterisk crashes, calls are lost, and that means - lost money. Are there any ideas? Regards, Atis -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Redundancy
- Original Message From: Atis Lezdins [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, September 25, 2007 2:11:10 PM Subject: Re: [asterisk-users] Asterisk Redundancy On 9/25/07, Philipp Kempgen [EMAIL PROTECTED] wrote: Adrian Marsh wrote: I'm interested in how people are clustering Asterisk, if that's possible, or how you might be achieving a redundant solution. I've a single Asterisk server driving the company. Its well backed-up, and I've a cloned machine that (in theory) with a DNS change could take over operations. However I'd like to achieve something more automated if possible.. Maybe my post at http://lists.digium.com/pipermail/asterisk-users/2007-August/195339.html could provide you with some answers. Hi, This seems nice way of sharing settings, however it wouldn't take over calls in progress. For us, currently the greatest problem is that whenever Asterisk crashes, calls are lost, and that means - lost money. Are there any ideas? You might want to take Asterisk out of the media path then. If it crashes, calls will stay up, although your CDR's will be screwed. If screwed CDR's still means lost money... your still screwed! Doug. Pinpoint customers who are looking for what you sell. http://searchmarketing.yahoo.com/___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dutch Number for Inbound
On 16:40, Tue 25 Sep 07, Dean Collins wrote: A friend of mine just sent me this email - he is looking for an IAX inbound service in Holland - any thoughts? Voip info only has Nadiz which seems to be SIP only. We use the following IAX providers with dutch telephone numbers in this order of preference: Speakup (http://www.speakup.nl) 12connect (http://www.12connect.com) voop (http://www.voop.com) -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called users? ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Grandstream GXP2020 / 2000
Has somebody experiences with the Grandstream GXP2020 / 2000 phones in a business graded installation I'm afraid I can't give you as positive feedback as you've had from other posters. I did quite a few installations with GXP2000s about 18 months ago, and they've caused us nothing but problems. Firmware started off as truly abysmal - half the features that the phones were sold as supporting didn't exist, and it wasn't until the more recent firmware that they were added. In the early firmware the speakerphone was practically unusable. In later versions they sorted out the echo problem by making it quieter, so it was still unusable in anything apart from a quiet single office. There was also an issue where 2 versions of the hardware were out there and a firmware update managed to kill the LCD display on a whole raft of phones. We upgraded the 3 offices left that hadn't replaced them with something else in one go. 2 of them worked fine, I had calls the following day from the third office saying none of their phones had any display. They've mostly fixed it in very recent releases, but I still have the odd one or two phones out there where the display just vanishes from time to time for no reason. Ignoring the firmware trials and tribulations for a moment, one fact still remains: the handset feels cheap and tacky, and compared to the Linksys SPA942 call quality is noticeably inferior (even LAN-LAN using g711). Here in the UK, the SPA942 is only about 10GBP more than the GXP2000 which makes it a much better choice. 2 of the 3 offices which had GXP2000s have replaced them with SPA942s over the last 6 months. The final one will be replacing the GXPs in a few weeks when they move office. It'll be like a support weight lifted off my back when we finally get rid of them all. [if anyone in the UK wants some second-hand GXP2000s I have quite a few, about 18 months old, in good condition :-) ] Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited For full contact details visit http://www.minotaur.it This email is made from 100% recycled electrons ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Point-to-Point SIP link without registration
Run it with a stock OpenSER installation that will accept registrations and acknowledge them with a 200 OK, but not actually do anything with them. I'm trying to avoid a PC at all in this scenario. If at all possible, I want an ATA at one end and a SIP phone at the other, no other hardware involved. Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited For full contact details visit http://www.minotaur.it This email is made from 100% recycled electrons ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF dropping digits
Hmm, this seems to describe my problem - Thanks, Bart -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tony Mountifield Sent: Tuesday, September 25, 2007 6:41 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] DTMF dropping digits In article [EMAIL PROTECTED], Barton Fisher [EMAIL PROTECTED] wrote: We have a Te410P with 3 Telco T1's (D4 SF ) with DID's (non-PRI). ANI DNIS is received in-band DTMF in a format such as *7145551212*8002* What happens when there are 30 or more calls, asterisk might see is DNIS = 802 or ANI = 4551212 for examples, where parts of the numbers are dropped. All the traffic arrives into a simple IVR script where a message is played. We are using Asterisk 1.2 and Server is 2.8 Dual Xeon SuperMicro with 2 GB RAM. Any clues what I can do to fix this? Try applying the patch at http://bugs.digium.com/view.php?id=10535 Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Configure one call per line on Cisco 7941/7961
Anyone aware of how to configure one call per line on a Cisco 7941/7961? The default behaviour is to have two calls per line button, and this is confusing for some of my users so I'd like to be able to have the 2nd call ring the second line button, rather than being shared with the first. I'm hoping this is something that is configurable in the XML or on the phone UI. Thanks Gary ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Point-to-Point SIP link without registration
You can do this with any of the Linksys SPA series ATA's or phones, just set Make Call Without Reg and Ans Call Without Reg to no. -- Eric Chamberlain, CISSP Chief Technical Officer Voxilla - http://voxilla.com/ -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Chris Bagnall Sent: Tuesday, September 25, 2007 11:34 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Point-to-Point SIP link without registration Greetings list, I need to set up a point to point SIP connection between two devices without either of them registering with a registrar/proxy/etc. at all. The devices I've tested so far all seem to insist on having a registration before they'll make or take calls. One of the devices needs to be an ATA with an FXO port (e.g. Sipura/Linksys SPA-3000/3102), the other device can be either an ATA or a SIP Phone. Does anyone have any hardware recommendations that'll work in this scenario? Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited For full contact details visit http://www.minotaur.it This email is made from 100% recycled electrons ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [on-asterisk] Configure one call per line on Cisco 7941/7961
Gary, if you register multiple lines with the same SIP credentials the phone will do rollover and take care of it. (2nd call comes in on L2, etc.) - dbc. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gary T. Giesen Sent: September-25-07 6:37 PM To: [EMAIL PROTECTED]; asterisk-users@lists.digium.com Subject: [on-asterisk] Configure one call per line on Cisco 7941/7961 Anyone aware of how to configure one call per line on a Cisco 7941/7961? The default behaviour is to have two calls per line button, and this is confusing for some of my users so I'd like to be able to have the 2nd call ring the second line button, rather than being shared with the first. I'm hoping this is something that is configurable in the XML or on the phone UI. Thanks Gary - To unsubscribe, e-mail: [EMAIL PROTECTED] For additional commands, e-mail: [EMAIL PROTECTED] ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Redundancy
[snip] http://lists.digium.com/pipermail/asterisk-users/2007-August/195339.html could provide you with some answers. Hi, This seems nice way of sharing settings, however it wouldn't take over calls in progress. For us, currently the greatest problem is that whenever Asterisk crashes, calls are lost, and that means - lost money. Are there any ideas? You might want to take Asterisk out of the media path then. If it crashes, calls will stay up, although your CDR's will be screwed. If screwed CDR's still means lost money... your still screwed! Nop, i can't stay out of media path, as there are essential features depending on it - hell, that's why i need asterisk - transfers, chanspy, monitoring.. Of course in case of crash - monitoring and CDR can be lost - that would be minor problem comparing to lost calls. I'm thinking about some mechanism how asterisk could communicate with second asterisk and report all state operations made with SIP. So if asterisk fails, redundancy asterisk performs IP takeover and continues. Unfortunately my SIP knowledge is nearly minimal (as are my C skills), and i don't have any ideas how to implement this. A simplest method could be something like SIP proxy, that sends calls to asterisk, but if asterisk stops responding, it plays some message and tries to send call to redundancy server - however then problem can occur with redundancy server. And this would have some major drawbacks - calls wouldn't be matched to corresponding agents in queue. Hmm, thinking a bit more about topic - maybe redundancy mechanism would have enough to keep state of channels, bridges, and corresponding dialplan location (assuming that config is identical). Too much of duplicating everything would mean that second asterisk could have the same crash. Regards, Atis -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Redundancy
A little off topic, but SipX has built in redudancy. if it is so important to you, you should have a look. On 9/25/07, Atis Lezdins [EMAIL PROTECTED] wrote: [snip] http://lists.digium.com/pipermail/asterisk-users/2007-August/195339.html could provide you with some answers. Hi, This seems nice way of sharing settings, however it wouldn't take over calls in progress. For us, currently the greatest problem is that whenever Asterisk crashes, calls are lost, and that means - lost money. Are there any ideas? You might want to take Asterisk out of the media path then. If it crashes, calls will stay up, although your CDR's will be screwed. If screwed CDR's still means lost money... your still screwed! Nop, i can't stay out of media path, as there are essential features depending on it - hell, that's why i need asterisk - transfers, chanspy, monitoring.. Of course in case of crash - monitoring and CDR can be lost - that would be minor problem comparing to lost calls. I'm thinking about some mechanism how asterisk could communicate with second asterisk and report all state operations made with SIP. So if asterisk fails, redundancy asterisk performs IP takeover and continues. Unfortunately my SIP knowledge is nearly minimal (as are my C skills), and i don't have any ideas how to implement this. A simplest method could be something like SIP proxy, that sends calls to asterisk, but if asterisk stops responding, it plays some message and tries to send call to redundancy server - however then problem can occur with redundancy server. And this would have some major drawbacks - calls wouldn't be matched to corresponding agents in queue. Hmm, thinking a bit more about topic - maybe redundancy mechanism would have enough to keep state of channels, bridges, and corresponding dialplan location (assuming that config is identical). Too much of duplicating everything would mean that second asterisk could have the same crash. Regards, Atis -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Guilherme Loch Góes MSN:[EMAIL PROTECTED] (48) 99115299 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk-users Digest, Vol 38, Issue 83
Hi! I am proving Asterisk 1.2.24 in realtime with MySQL 5.0.27 using Idefisk softphones. I followed the steps of how to of voip-org but always have this error: Sep 25 20:29:07 WARNING[12000]: res_config_mysql.c:360 update_mysql: MySQL RealTime: Failed to query database. Check debug for more info. Sep 25 20:29:07 WARNING[12000]: res_config_mysql.c:360 update_mysql: MySQL RealTime: Failed to query database. Check debug for more info. Sep 25 20:29:07 WARNING[12000]: res_config_mysql.c:360 update_mysql: MySQL RealTime: Failed to query database. Check debug for more info. Sep 25 20:29:07 NOTICE[12000]: chan_iax2.c:5252 register_verify: Host 127.0.0.1 failed MD5 authentication for '101' (9a43a82001dfa49d84e8facb765f7de2 != 31610d29241e861816b83998501ee223) I configure extconfig.conf as: [settings] iaxusers = mysql,asterisk,iax_buddies iaxpeers = mysql,asterisk,iax_buddies sipusers = mysql,asterisk,sip_buddies sippeers = mysql,asterisk,sip_buddies res_mysql.conf as: [general] dbhost = localhost dbname = asterisk dbuser = root dbpass = asterisk dbport = 3306 dbsock = /var/lib/mysql/mysql.sock My table as: CREATE TABLE iax_buddies ( name varchar(30) primary key NOT NULL, username varchar(30), type varchar(6) NOT NULL, secret varchar(50), callerid varchar(100), context varchar(100), host varchar(31) NOT NULL default 'dynamic', disallow varchar(100), allow varchar(100) ); I'm running asterisk on Fedora 6. Plz help thanks in advance Renzzo ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Grandstream GXP2020 / 2000
Has somebody experiences with the Grandstream GXP2020 / 2000 phones in a business graded installation (with really traffic on not 3 calls a day ;-) ) Of course with Asterisk PBX (1.4.1 or 1.4.11 or 1.4 I have an installation right now in a real estate/mortgage company office with 36 GXP2000 phones. Average call volume is currently only about 150-200 calls per day but the number is climbing rapidly as they add more agents/loan officers. The latest firmware (Beta: 1.1.4.22) is a huge improvement over 1.1.1.14 though 1.1.1.14 is the current release firmware. Sadly some of the firmware loads we've tested have been horrible! The speaker phone is greatly improved. Call forwarding was an issue for several loads if using Asterisk 1.4 or later. Seems the SIP 302 message coming from the phone was corrupted. I had to hack chan_sip.c as a work around because this was a feature the had worked using 1.2 and was promised with the new system. Version 1.1.4.18 finally fixed that issue so the hack isn't necessary any more. The biggest complaint I have is the method of creating a config files for the phones. Unlike a Polycom which allows you to configure the phone using an XML file, the GXP requires you to create a text file with the configuration settings and then compile that file with their software. Additionally, if you perform a factory reset on the phone, it tries to connect to fm.grandstream.com/gs to update it's firmware load. So we are forced to run a caching name server with that address pointing to our own local server. (Don't even bother trying to tell me that all I have to do is change it in the web interface. After a factory reset, or a nice ESD zap which seems to nearly always result in a factory reset, the default address is back.) We have approached the configuration issue several different ways. The current method is using a MySQL database. We built the database and then modified the HTML from the phones web configuration to use it to update the database. We use a cron to monitor the last update time and generate a new set of config files once the database has been updated. If you only have a small installation or very little turnover, our previous method of using a text file for the database and a perl script to update the files is probably sufficient. While I haven't gotten any complaints about the cheap toy like feel, I think this is mostly due to lack of experience on the part of my users. With the GXP being the only VoIP phone they have used, they do not have a basis for comparison. The original quote offered Polycom, Aastra, Snom, and Linksys phones. The GXP was chosen strictly by price since the price difference saved them over $1000. I now demonstrate the phones on a portable system to allow the customer to see and feel the difference in the phones. I have also increased the price of the GXP phones I sell. Between these two measures I don't sell as many GXP phones. I feel the increase in price was justified based on the additional work I've had in using the GXP phones. Since the bugs are mostly worked out now, these should be more profitable for me in the future. John ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [on-asterisk] Configure one call per line on Cisco 7941/7961
David, Yes, I'm aware of that, but unfortunately it does two calls on each line appearance (button), so the first two calls go on line 1, and the third will appear on line 2. I'd like to limit it to 1 call per line. Any ideas? Gary On 9/25/07, David Cook [EMAIL PROTECTED] wrote: Gary, if you register multiple lines with the same SIP credentials the phone will do rollover and take care of it. (2nd call comes in on L2, etc.) - dbc. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gary T. Giesen Sent: September-25-07 6:37 PM To: [EMAIL PROTECTED]; asterisk-users@lists.digium.com Subject: [on-asterisk] Configure one call per line on Cisco 7941/7961 Anyone aware of how to configure one call per line on a Cisco 7941/7961? The default behaviour is to have two calls per line button, and this is confusing for some of my users so I'd like to be able to have the 2nd call ring the second line button, rather than being shared with the first. I'm hoping this is something that is configurable in the XML or on the phone UI. Thanks Gary - To unsubscribe, e-mail: [EMAIL PROTECTED] For additional commands, e-mail: [EMAIL PROTECTED] ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP Panel?
Dear List, Has anyone found or written a status panel application, windows or linux, that uses SIP notifies and subscriptions, to gather the status of SIP extensions from Asterisk? And displsy nicely on a GUI? -- Terence C. Giufre-Sweetser Technical Support Network Engineering SkyMesh Pty Ltd Licensed Telecommunications Carrier ABN 62 113 609 439 47 Baxter Street FORTITUDE VALLEY Q 4006 Support Hotline 1300 759 637 Support Hours 8:00 am – 8:00 pm Monday - Friday 10:00 am – 4:00 pm Weekends Public Holidays ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Grandstream GXW-4008
I'm trying to use a GXW-4008 for the first time to provide simple POTS. Is anyone using it? How about samples of SIP.CONF and EXTENSIONS.CONF? Do you have advice for configuring the GXW-400x for this application? How long a local loop will it support on the FXS ports? When I started to configure the unit, I was able to connect via the WAN port. Now I'm unable to connect to it using the IP address supplied by '02' in the IVR. '12' indicates that the WAN access is enabled. The 'ready' light is flashing; I don't see anything in the manual that indicates what this means. I've reset to factory settings using the reset button on the back, but that didn't change the situation. --Don Don Kelly PCF Corp Real Support for your Virtual Office 651 842-1000 888 Don Kell(y) 651 842-1001 fax ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Yikes! Polycom 501 chokes on BootRom 4.0.0?
One more thing i noticed today, with SIP 2.2 and Polycom 601 i wasnt able to enable buddy watch to use with hints. I'll spend more time on it later to see what is up with that. On 9/25/07, Mike [EMAIL PROTECTED] wrote: I am having a similar issue with 4.0.0. Mine is that it doesn't get any DHCP address (gets stuck waiting for an address). I fixed it by going back one to the previous bootrom version, worked like a charm. Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dave Fullerton Sent: Tuesday, September 25, 2007 08:49 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Yikes! Polycom 501 chokes on BootRom 4.0.0? Doug wrote: I was progressively upgrading this phone from 3.1.2 to 3.2.3, then to 4.0.0. v3.2.3 worked fine, but when I went to 4.0.0 (Even adding the more specific 2345-11500-040.bootrom.ld), it won't run, and just keeps rebooting. Now I've got a really nice doorstop unless someone knows how to get out of this predicament. Help! 0925003705|cfg |3|00|Beginning to provision phone dns |3|00|DNS 0925003705|lookup for 'somedomain.com'(66.16.26.106) TTL=83485 copy |3|00|'ftp://someuser:[EMAIL PROTECTED]/2345-11500-040.bootrom.ld' from 'somedomain.com(66.16.26.106)' 0925003706|cfg |3|00|Image 2345-11500-040.bootrom.ld has not changed 0925003706|copy |3|00|Download of '2345-11500-040.bootrom.ld' succeeded on attempt 1 (addr 1 of 1) 0925003706|cfg |3|00|Downloaded bootROM is identical to current 0925003706|version 4.0.0 copy |3|00|'ftp://someuser:[EMAIL PROTECTED]/0004f210.cfg' from 'somedomain.com(66.16.26.106)' 0925003707|copy |3|00|Download of '0004f210.cfg' succeeded on attempt 1 (addr 1 of 1) 0925003708|cfg |5|00|Could not get the list of CONFIG_FILES cfg 0925003708||5|00|Could not get the list of MISC_FILES 0925003709|cfg |5|00|Couldn't get parameter APP_FILE_PATH cfg 0925003709||3|00|Unspecified error occured with downloaded application image 0925003709|app1 |6|00|Error in saving application. 0925003709|app1 |6|00|Uploading boot log, time is TUE SEP 25 00:37:10 0925003709|2007 I've upgraded my 501 to bootrom 4.0.0. It did reboot and reformat the filesystem about three times in a row before it finally finished but it did work for me. I'm still using SIP 2.1.2 though. Don't know if that information helps any. -Dave ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] swift.conf - cepstral voice quality adjustment options
There have been a number of instances where recent changes in the * code have led to a degradation of TTS in the 1.4 releases. I have no idea whether this is relevant to ABE in general or the version you're running. However, for a number of us the fix was to edit app_swift.c (version 2.0rc1 from http://www.mezzo.net/asterisk/app_swift.html) and change the line const int framesize = 160 * 2; (I *think* that's what it was originally) to const int framesize = 20; Recompile app_swift, reload and all was good :) Then again, this may have no relevance to ABE. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
