Re: [asterisk-users] Re: checking 'voicemail externally - doesn't work
On Mon, 2006-10-23 at 00:48 -0500, Eric ManxPower Wieling wrote:
Voicemail allows the caller to leave voicemail. Voicemailmain allows
you to check your voicemail.
I got this one.
1.0.x Asterisk mailbox options were put as a prefix to the mailbox,
such
as Voicemail(u11) would play the unavailable message to the caller.
1.2 (I think) changed this to make it more like all the other
applications, i.e., use a , or | before the options.
Though what option am I suppose to pass it.
The process seems to me correct, when I get-in to disa-access I have
access to voicemail extension 1000 (otherwise it wouldn't let me dial
ext. 1000; when I dial it it asking me for mailbox number and password,
except that password is not recognized; even tough I see it from the
command line that the correct password 123 was entered. So I don't
understand why isn't it accepting it?
[voicemail]
exten = 1000,1,NoCDR()
exten = 1000,2,Answer()
exten = 1000,3,VoicemailMain(${CALLERIDNUM})
[disa-access]
include = tollfree
include = voicemail
--
#Joseph
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Re: [asterisk-users] Audiocodes MP-20x
Hi Has anyone used the AudioCodes MP-20x?I've been testing this for 3 weeks now. No problems so far. This gateway has many features including IPSec and is not that expensive. RegardsAndrew ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Realtime... Help Me!!!
Great! Thanks for your aid... I spend a lot of day around this problem... Now realtime load returns data! - Original Message - From: Tijl Van den Broeck [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, October 23, 2006 7:52 AM Subject: Re: [asterisk-users] Asterisk Realtime... Help Me!!! Thanks alot! Indeed it was the 3th solution, I changed sipusers = odbc,MySQL-asterisk,sip_buddies sippeers = odbc,MySQL-asterisk,sip_buddies into sipusers = odbc,mysql2,sip_buddies sippeers = odbc,mysql2,sip_buddies And realtime load sipusers username 1006 now returns data :-) greets Tijl Van den Broeck On 10/23/06, Benjamin Jacob [EMAIL PROTECTED] wrote: Make additional checks : 1) ensure u've unixodbc, unixodbc-devel installed, use this command rpm -qa | grep -i unixodbc MUST see unixodbc and unixodbc-devel in the output!!!, else get unixodbc and unixodbc-devel(am kinda guessing u do have that perfect). 2) /etc/odbcinst.ini and /etc/odbc.ini should be correct. cross check 3) Aaahh.. revelation!! I think, I know where you've gone wrong. In your res_odbc.conf , you have given the database context as mysql(see [mysql]). This should be the same as the 2nd argument in ur extconfig.conf line for realtime for your sipusers. i.e. it should be sipusers = odbc,mysql,sipusers instead of sipusers = odbc,asterisk,sipusers This should work fine. If it doesn't, paste your odbc.ini and odbcinst.ini files as well over here. or give me ssh login access to your machine.(dont wory, wont mess up ur machine). cheerz - Ben. Maurizio Pederneschi wrote: These are my conf file: res_odbc.conf ;;; odbc setup file ; ENV is a global set of environmental variables that will get set. ; Note that all environmental variables can be seen by all connections, ; so you can't have different values for different connections. [ENV] INFORMIXSERVER = my_special_database INFORMIXDIR = /opt/informix ; All other sections are arbitrary names for database connections. ;[asterisk] ;enabled = yes ;dsn = asterisk ;;username = myuser ;;password = mypass ;pre-connect = yes [mysql] enabled = yes dsn = MySQL-asterisk username = root password = pre-connect = yes --- - - extconfig.conf ; ; Static and realtime external configuration ; engine configuration ; ; Please read doc/README.extconfig for basic table ; formatting information. ; [settings] ; ; Static configuration files: ; ; file.conf = driver,database[,table] ; ; maps a particular configuration file to the given ; database driver, database and table (or uses the ; name of the file as the table if not specified) ; ;uncomment to load queues.conf via the odbc engine. ; ;queues.conf = odbc,asterisk,ast_config ; ; The following files CANNOT be loaded from Realtime storage: ; asterisk.conf ; extconfig.conf (this file) ; logger.conf ; ; Additionally, the following files cannot be loaded from ; Realtime storage unless the storage driver is loaded ; early using 'preload' statements in modules.conf: ; manager.conf ; cdr.conf ; rtp.conf ; ; ; Realtime configuration engine ; ; maps a particular family of realtime ; configuration to a given database driver, ; database and table (or uses the name of ; the family if the table is not specified ; ;example = odbc,asterisk,alttable ;iaxusers = odbc,asterisk ;iaxpeers = odbc,asterisk sipusers = odbc,asterisk,sipusers ;sippeers = odbc,asterisk voicemail = odbc,asterisk ;extensions = odbc,asterisk ;queues = odbc,asterisk ;queue_members = odbc,asterisk extensions = odbc,asterisk,extensions --- - This is my table sipusers | id | name | username | context | host| port | secret | allow | ipaddr | type | password | | 1 | pippo| pippo| tutorial | dynamic | | password | g729;ilbc;gsm;ulaw;alaw | NULL | friend | password | | 2 | testAsterisk | testAsterisk | tutorial | dynamic | | password | g729;ilbc;gsm;ulaw;alaw | NULL | friend | password | --- - This is the output of the realtime load command: realtime load sipusers name pippo No rows found matching search criteria. Thank's Maury - Original Message - From: Benjamin Jacob [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, October 20, 2006 12:39 PM Subject: Re: [asterisk-users]
Re: [asterisk-users] Asterisk Realtime... Help Me!!!
Alls well that ends well !!! :-) Maurizio Pederneschi wrote: Great! Thanks for your aid... I spend a lot of day around this problem... Now realtime load returns data! - Original Message - From: Tijl Van den Broeck [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, October 23, 2006 7:52 AM Subject: Re: [asterisk-users] Asterisk Realtime... Help Me!!! Thanks alot! Indeed it was the 3th solution, I changed sipusers = odbc,MySQL-asterisk,sip_buddies sippeers = odbc,MySQL-asterisk,sip_buddies into sipusers = odbc,mysql2,sip_buddies sippeers = odbc,mysql2,sip_buddies And realtime load sipusers username 1006 now returns data :-) greets Tijl Van den Broeck On 10/23/06, Benjamin Jacob [EMAIL PROTECTED] wrote: Make additional checks : 1) ensure u've unixodbc, unixodbc-devel installed, use this command rpm -qa | grep -i unixodbc MUST see unixodbc and unixodbc-devel in the output!!!, else get unixodbc and unixodbc-devel(am kinda guessing u do have that perfect). 2) /etc/odbcinst.ini and /etc/odbc.ini should be correct. cross check 3) Aaahh.. revelation!! I think, I know where you've gone wrong. In your res_odbc.conf , you have given the database context as mysql(see [mysql]). This should be the same as the 2nd argument in ur extconfig.conf line for realtime for your sipusers. i.e. it should be sipusers = odbc,mysql,sipusers instead of sipusers = odbc,asterisk,sipusers This should work fine. If it doesn't, paste your odbc.ini and odbcinst.ini files as well over here. or give me ssh login access to your machine.(dont wory, wont mess up ur machine). cheerz - Ben. Maurizio Pederneschi wrote: These are my conf file: res_odbc.conf ;;; odbc setup file ; ENV is a global set of environmental variables that will get set. ; Note that all environmental variables can be seen by all connections, ; so you can't have different values for different connections. [ENV] INFORMIXSERVER = my_special_database INFORMIXDIR = /opt/informix ; All other sections are arbitrary names for database connections. ;[asterisk] ;enabled = yes ;dsn = asterisk ;;username = myuser ;;password = mypass ;pre-connect = yes [mysql] enabled = yes dsn = MySQL-asterisk username = root password = pre-connect = yes --- - - extconfig.conf ; ; Static and realtime external configuration ; engine configuration ; ; Please read doc/README.extconfig for basic table ; formatting information. ; [settings] ; ; Static configuration files: ; ; file.conf = driver,database[,table] ; ; maps a particular configuration file to the given ; database driver, database and table (or uses the ; name of the file as the table if not specified) ; ;uncomment to load queues.conf via the odbc engine. ; ;queues.conf = odbc,asterisk,ast_config ; ; The following files CANNOT be loaded from Realtime storage: ; asterisk.conf ; extconfig.conf (this file) ; logger.conf ; ; Additionally, the following files cannot be loaded from ; Realtime storage unless the storage driver is loaded ; early using 'preload' statements in modules.conf: ; manager.conf ; cdr.conf ; rtp.conf ; ; ; Realtime configuration engine ; ; maps a particular family of realtime ; configuration to a given database driver, ; database and table (or uses the name of ; the family if the table is not specified ; ;example = odbc,asterisk,alttable ;iaxusers = odbc,asterisk ;iaxpeers = odbc,asterisk sipusers = odbc,asterisk,sipusers ;sippeers = odbc,asterisk voicemail = odbc,asterisk ;extensions = odbc,asterisk ;queues = odbc,asterisk ;queue_members = odbc,asterisk extensions = odbc,asterisk,extensions --- - This is my table sipusers | id | name | username | context | host| port | secret | allow | ipaddr | type | password | | 1 | pippo| pippo| tutorial | dynamic | | password | g729;ilbc;gsm;ulaw;alaw | NULL | friend | password | | 2 | testAsterisk | testAsterisk | tutorial | dynamic | | password | g729;ilbc;gsm;ulaw;alaw | NULL | friend | password | --- - This is the output of the realtime load command: realtime load sipusers name pippo No rows found matching search criteria. Thank's Maury - Original Message - From: Benjamin Jacob [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, October 20, 2006 12:39 PM Subject: Re: [asterisk-users] Asterisk Realtime... Help Me!!! Maurizio Pederneschi wrote: Hi, i have
Re: [asterisk-users] VoicePulse Connect 4 Channel Limit?
So, What´s your recommendation for a production environment? I waslooking for good prices, good voice quality for USA Origination and I´d like to hear about good experiences PSTN. Just can't beat the quality :-) Wait, you said good prices. Sorry. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Audiocodes MP-20x
On 10/23/06, Andrew Nowrot [EMAIL PROTECTED] wrote: I've been testing this for 3 weeks now. No problems so far. This gateway has many features including IPSec and is not that expensive. Appreciate if you can post the sample configs to wiki or to the list. There is no information about configuring Audiocodes with asterisk. raj ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Compiling H323 channel Asterisk 1.4.Beta3?
Is it possible to compile the h323 channel before installing the full asterisk package as mentioned within the README Within the Asterisk/channels/h323 Directory It compiles after asterisk has been installed, but no chan_h323.so has been created within the channels directory Been able to compile all variants of h323 on previous asterisk installations.. Any Ideas Thanks again ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] accountcode and amaflags?
Any more ideas, esp from guys whove used this in their setp? Benjamin Jacob wrote: Giovanni, Appreciate your lines mate. But, Ive already read those, all over the net. my qs inline : amaflags : Categorization for CDR records. Choices are default, omit, billing, documentation and choices are defaul, omit, billing, documentation wot r these categories??wot decides these categories? accountcode : string : Users may be associated with an accountcode (billing purpose) hmm.. ive seen in quite a few places, where the pin collected is stored as the accountcode... wot duz that mean? anyway, can you give me an example of wot the association means?am a lil slow.. Cheers, Giovanni 2006/10/19, Benjamin Jacob [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]: Hello ppl, Can someone explain to me the meaning and use of the variables accountcode and amaflags in sip.conf,etc. Googled, voip-infoed, wikied, etc for it. Couldnt get much of it. I know, they are billing related, but not much beyond that. Any ideas? cheerz ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Why does it take at least 4 flipping days before asterisk tries to resolve a provider?
After a reboot, asterisk is usually too much in a hurry to try and resolve my iax/sip providers. Asterisk starts before the internet connection is up and dns is working. Then asterisk just waits, and waits and waits and waits even longer before ever trying to revolve any voip provider again. And all this time calls are flowing out through the very expensive PSTN. And then people say nightly asterisk restarts are not a good idea ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Primary D-Channel on span 2 down
Hello All! At start of an asterisk I see the following: == Primary D-Channel on span 2 up == Primary D-Channel on span 2 down Oct 23 10:26:25 WARNING[9515]: chan_zap.c:2287 pri_find_dchan: No D-channels available! Using Primary channel 47 as D-channel anyway! == Primary D-Channel on span 2 up == Primary D-Channel on span 2 up == Primary D-Channel on span 2 up == Primary D-Channel on span 2 up . What can be to it the reason? begin:vcard fn:Eugeniy Khvastunov n:Khvastunov;Eugeniy org:Digma;IT adr:;;;Kharkov;Kh;;Ukraine email;internet:[EMAIL PROTECTED] title:System Administrator tel;work:+380675745646 tel;cell:+380504063116 version:2.1 end:vcard ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Why does it take at least 4 flipping days before asterisk tries to resolve a provider?
Remco, Asterisk starts before the internet connection is up and dns is working. knip And then people say nightly asterisk restarts are not a good idea Why is your asterisk startup script running before networking has been setup? Asterisk has the same networking dependencies as apache, so I start it around the same time using the same priority as apache and as far as I know networking should work at that time or not at all, not somewhere in between. pebkac? -- Andreas SikkemaBBeyond Software EngineerPlaneetbaan 4 +31 (0)23 70743422132 HZ Hoofddorp ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Can anyone help? Why does One-Touch record mute/disconnect callif not dialed quick enough?
Hi, Any suggestions to below problem? Thanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jamie Heckford Sent: 17 October 2006 21:48 To: asterisk-users@lists.digium.com Subject: [asterisk-users] FW: Why does One-Touch record mute/disconnect callif not dialed quick enough? Hi List, Have an odd problem with the one-touch record on asterisk 1.2.11. All works ok, however one of our users today discovered if he is a bit slow hitting the 1 key after he presses *, the call seems to stay connected but its almost like it is muted. Haven't figured out the delay yet but it seems to be if the 1 is not pressed within 1-2 secs this occurs. Any suggestions? I tried setting: disconnect = *0 in features.conf in the hope this would solve it but no luck. I am using Polycom SPIP 301 handsets and can't see anything obvious on these either/ Thanks in advance for any help! Kind regards Jamie Heckford Technical Consultant Interfuture Systems Ltd Kemps Farm Business Park, London Road, Balcombe, West Sussex RH17 6JH E-MAIL DISCLAIMER: This e-mail is intended for the addressee named above only, and may be covered by legal privilege and/or protected by law. If you are not the intended recipient please notify the sender immediately, and in the meantime do not disclose the contents to any other person nor use, copy or store the e-mail in any medium. As communications via the Internet are not secure Interfuture Systems Ltd can accept no liability if this e-mail is accessed by third parties during the course of transmission or is modified or amended in any way following despatch. Any views or opinions expressed within this e-mail are solely those of the sender, and do not necessarily represent those of Interfuture Systems Ltd unless otherwise specifically stated. Although Interfuture Systems Ltd has taken every reasonable precaution to ensure that any attachment to this e-mail has been checked for viruses, it is strongly recommended that you carry out your own virus check before opening any attachment, as we cannot accept liability for any damage sustained as a result of software virus infection. Interfuture Systems Ltd reserves the right and senders of messages shall be taken to consent to the monitoring and recording of e-mails addressed to members of the firm. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Can't change Zaptel driver from FCC mode
Hi, I'm using Asterisk with a Digium TDM10B FXO card and it's driving me
nuts. I'm based in the UK and have echo problems and need to switch the
driver from FCC mode to UK mode.
I've tried modprobe zaptel and modprobe wctdm opermode=UK and the ztcfg. I
get no error messages but when I reboot it still comes up as FCC mode?
What am I doing wrong?
Neil
safeharbour IT Ltd
Your IT Department
tel: 0845 644 3607
fax: 0845 867 2891
mob: 07812 114784
voip: [EMAIL PROTECTED]
email: [EMAIL PROTECTED]
web: www.safeharbourit.co.uk
The information in this e-mail is confidential and may be legally
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business.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Todd- Asterisk
Sent: 20 October 2006 19:29
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] getting DID info..
Thanks for the help Jerry - I'm getting closer, but still no luck...
Now, I hear the lady say S. I think what is happening is that the GoTo
command is setting the extension to 's' when it transfers control to the
context defined in the IAX.conf -where I have the trunk line defined...
exten = h,1,Hangup
exten = s,n,Answer
exten = s,n,Wait(1)
exten = s,n,SayAlpha(${EXTEN})
It is my impression that the EXTEN variable is used as the internal
extension - not the incoming DID number, but you seem pretty confident so I
must be wrong. What Im looking to do is a FOP pop-up
with the DID number and caller ID number in it... I'll tie that
into a web-based database...
Here's my full log file..
Oct 20 14:23:42 VERBOSE[5387] logger.c: -- Accepting
AUTHENTICATED call from 204.11.194.34:
requested format = ulaw,
requested prefs = (),
actual format = ulaw,
host prefs = (ulaw|alaw|gsm),
priority = mine
Oct 20 14:23:42 VERBOSE[5862] logger.c: -- Executing Set
(IAX2/204.11.194.34:4569-4, LOOPCOUNT=0) in new stack
Oct 20 14:23:42 VERBOSE[5862] logger.c: -- Executing Set
(IAX2/204.11.194.34:4569-4, __DIR-CONTEXT=default) in new stack
Oct 20 14:23:42 VERBOSE[5862] logger.c: -- Executing Answer
(IAX2/204.11.194.34:4569-4, ) in new stack
Oct 20 14:23:42 VERBOSE[5862] logger.c: -- Executing Wait
(IAX2/204.11.194.34:4569-4, 1) in new stack Oct 20 14:23:43 DEBUG[5387]
chan_iax2.c: Ooh, voice format changed to 4
Oct 20 14:23:43 VERBOSE[5862] logger.c: -- Executing SayAlpha
(IAX2/204.11.194.34:4569-4, s) in new stack Oct 20 14:23:43 DEBUG[5862]
channel.c: Scheduling timer at 160 sample intervals
Oct 20 14:23:43 VERBOSE[5862] logger.c: -- Playing 'letters/
s' (language 'en')
DID is the inbound call number.
The is notation for CallerID name, that won't help.
s is the start extension. setting it to FROM_DID makes no sense.
(This is the extention that starts in this context; it is a default,
if the context is started without an extension. (eg batphone or called
from another
context))
FROM_DID=${EXTEN} gets you the right number.
However, SayNumber is looking for a SINGLE digit. Your
000-000- style number is overflow, and hence zero.
You have to parse the number to do this right.
If you aren't sure how, let me know, I might have a macro to do it.
Thanks,
J.
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Re: [Asterisk-Users] rxfax problem
On 10/20/06, Mohammad Shokuie [EMAIL PROTECTED] wrote: Anyways, let me take the most benefit as im sure you'd read this post, i have problem with the size of received page which is shrinked, can u give me a hint about this problem too :) This is probably the problem of the application that you use to view the TIFF file. FAX machines generate TIFF files with different horizontal and vertical resolution, and a lot of lazy programs do not check this correctly. I find that a quick 'tiff2pdf' conversion fixes things up very nicely :) Steve D ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Why does it take at least 4 flipping days before asterisk tries to resolve a provider?
On Mon, 2006-10-23 at 10:14 +0200, Andreas Sikkema wrote: Why is your asterisk startup script running before networking has been setup? Asterisk has the same networking dependencies as apache, so I start it around the same time using the same priority as apache and as far as I know networking should work at that time or not at all, not somewhere in between. I've seen this type of nonsense with newer versions of Suse. I've never bothered to find out why, just changed the priority manually to make sure it does what _I_ want. -- Dave Cotton [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Unicall Installation
Hi, Could anyone knows whatwent wrong with theerror below result of installation of libsupertone. [EMAIL PROTECTED] latest]# tar xvf libsupertone-0.0.2.tarlibsupertone-0.0.2/libsupertone-0.0.2/AUTHORSlibsupertone-0.0.2/Makefile.amlibsupertone-0.0.2/COPYINGlibsupertone-0.0.2/config/libsupertone-0.0.2/config/ltmain.shlibsupertone-0.0.2/config/missinglibsupertone-0.0.2/config/install-shlibsupertone-0.0.2/config/config.guesslibsupertone-0.0.2/config/depcomplibsupertone-0.0.2/config/config.sublibsupertone-0.0.2/configurelibsupertone-0.0.2/NEWSlibsupertone-0.0.2/libsupertone.speclibsupertone-0.0.2/ChangeLoglibsupertone-0.0.2/Makefile.inlibsupertone-0.0.2/supertone.clibsupertone-0.0.2/configure.inlibsupertone-0.0.2/libsupertone.hlibsupertone-0.0.2/INSTALLlibsupertone-0.0.2/supertone.hlibsupertone-0.0.2/libsupertone.spec.inlibsupertone-0.0.2/READMElibsupertone-0.0.2/supertone_tests.clibsupertone-0.0.2/config-h.inlibsupertone-0.0.2/aclocal.m4[EMAIL PROTECTED] latest]# ./configure --prefix=/usr/local/lib-bash: ./configure: No such file or directory[EMAIL PROTECTED] latest]# Help, pleeeaaassseee... Angel ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] SIP_HEADER function; what names are available?
Looking at the source code for Asterisk 1.2.7.1 (just what I've got
handy), it appears that the SIP_HEADER() function just parses the SIP
INVITE for whatever SIP *header* you specify - so:
a) there's no list of headers you can check for - it depends on the user
agent generating the request and
b) the request URI is not a SIP header, so you can't get to it using a
stock SIP_HEADER() function.
However, I suppose that there is nothing stopping you from hacking the
source for your Asterisk installation to provide access to the URI... In
chan_sip.c:func_header_read() you could do something like:
static char *func_header_read(struct ast_channel *chan, char *cmd, char
*data, char *buf, size_t len)
{
snip/
content = get_header(p-initreq, data);
if (ast_strlen_zero(content)) {
new
/* look for an experimental pseudo-header that allows us
access to the request URI */
/* but note that this is not a real header name! */
if (strcmp(data,
x-Asterisk-Request-URI-pseudo-header)==0)
{
ast_copy_string(buf, p-initreq.rlPart2, len);
ast_mutex_unlock(chan-lock);
return buf;
}
/new
ast_mutex_unlock(chan-lock);
return NULL;
}
snip/
}
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Ricardo Carvalho
Sent: 20 October 2006 17:51
To: kjcsb; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] SIP_HEADER function; what names
are available?
Any news on this thread? I also need to know the way to get
the R-URI from sip INVITE messages received by Asterisk,
through ${SIP_HEADER()}.
Thanks in advance,
Ricardo.
kjcsb wrote:
I have read the wiki about the SIP_HEADER function
(http://www.voip-
info.org/wiki/index.php?page=Asterisk+func+sip_header).
Where can I
get a list of the names that are available to be used with the
function e.g. TO is one name as in ${SIP_HEADER(TO)}.
What are the
others?
I would guess that you can check the RFC. Easier is to
turn on SIP
debug and see the INVITE packet yourself and
check the headers that you have with your equipment.
/Olle
Thanks but I don't know how to get the actual INVITE details (the
request URI?). For example I want to get
sip:[EMAIL PROTECTED]
SIP/2.0 from the following dialogue:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Record-Route: sip:147.202.nn.nnn;ftag=bf7eced18eb7271b;lr=on
Via: SIP/2.0/UDP 147.202.nn.nnn;branch=z9hG4bKe49c.21b320a3.0
Via: SIP/2.0/UDP 60.234.nnn.nnn;branch=z9hG4bK76bf3dec8d45b972
From: User sip:[EMAIL PROTECTED];tag=bf7eced18eb7271b
To: sip:[EMAIL PROTECTED]
etc
I can get Record-Route, Via, From, To etc but don't know how to get
the bit after the INVITE. Interestingly only the first Via
is returned
by ${SIP_HEADER(VIA)}.
I've tried R-URI, RURI, URI, ALL, *, blank.
Any advice appreciated.
Cameron
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Re: [asterisk-users] Unicall Installation
On Monday 23 October 2006 21:45, Angel Heart wrote: Hi, Could anyone knows what went wrong with the error below result of installation of libsupertone. [EMAIL PROTECTED] latest]# tar xvf libsupertone-0.0.2.tar [snip] libsupertone-0.0.2/aclocal.m4 [EMAIL PROTECTED] latest]# ./configure --prefix=/usr/local/lib -bash: ./configure: No such file or directory [EMAIL PROTECTED] latest]# Help, pleeeaaassseee... You probably shouldn't blindly follow instructions if you don't know what they do. ./configure should be running the script called configure in the current directory. Which, as the error message states, doesn't exist. You need to change into the correct directory (cd) before you execute the script. -- http://nicegear.co.nz New Zealand's VoIP Supplier ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unicall Installation
Hi, Thank you for your comment; Below was the result of ./configure checking how to run the C++ preprocessor... /lib/cppconfigure: error: C++ preprocessor "/lib/cpp" fails sanity checkSee `config.log' for more details.[EMAIL PROTECTED] libsupertone-0.0.2]# Please comment. Thanks again. - Original Message From: Hadley Rich [EMAIL PROTECTED]To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.comSent: Monday, October 23, 2006 5:01:16 PMSubject: Re: [asterisk-users] Unicall Installation On Monday 23 October 2006 21:45, Angel Heart wrote: Hi, Could anyone knows what went wrong with the error below result of installation of libsupertone. [EMAIL PROTECTED] latest]# tar xvf libsupertone-0.0.2.tar[snip] libsupertone-0.0.2/aclocal.m4 [EMAIL PROTECTED] latest]# ./configure --prefix=/usr/local/lib -bash: ./configure: No such file or directory [EMAIL PROTECTED] latest]# Help, pleeeaaassseee...You probably shouldn't blindly follow instructions if you don't know what they do../configure should be running the script called configure in the current directory. Which, as the error message states, doesn't exist. You need to change into the correct directory (cd) before you execute the script.-- http://nicegear.co.nzNew Zealand's VoIP Supplier___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can't change Zaptel driver from FCC mode
On Mon, Oct 23, 2006 at 09:34:10AM +0100, Neil Tancock wrote: Hi, I'm using Asterisk with a Digium TDM10B FXO card and it's driving me nuts. I'm based in the UK and have echo problems and need to switch the driver from FCC mode to UK mode. I've tried modprobe zaptel and modprobe wctdm opermode=UK and the ztcfg. I get no error messages but when I reboot it still comes up as FCC mode? What is the actual parameter? Any chance it is set elsewhere or that the module was already loaded? To check the current value: cat /sys/modules/wctdm/parameters/opermode Also verify you don't use the parameter _opermode. -- Tzafrir Cohen iax:[EMAIL PROTECTED]/tzafrir icq#16849755 mailto:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unicall Installation
On Mon, Oct 23, 2006 at 02:11:22AM -0700, Angel Heart wrote: Hi, Thank you for your comment; Below was the result of ./configure checking how to run the C++ preprocessor... /lib/cpp configure: error: C++ preprocessor /lib/cpp fails sanity check See `config.log' for more details. [EMAIL PROTECTED] libsupertone-0.0.2]# You don't have g++/gcc-c++ installed. You just need to install some packages. Which Linux distribution do you use? -- Tzafrir Cohen iax:[EMAIL PROTECTED]/tzafrir icq#16849755 mailto:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] astdb error, please help
Hello friends, I am getting this error:- Oct 23 15:47:22 WARNING[2124]: db.c:171 ast_db_put: Unable to put value '192.168.1.12:5060:300:15553695861:sip:[EMAIL PROTECTED]:5060' for key '23' in family 'SIP/Registry I have no idea what it means. Please tell me what could be the problem. With warm regards. Vivek J. Joshi. [EMAIL PROTECTED] Trikon electronics Pvt. Ltd. All science is either physics or stamp collecting. -- Ernest Rutherford ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] SIP_HEADER function; what names are available?
Minor update - use the following:
if (strcasecmp(data,
x-Asterisk-Request-URI-pseudo-header)==0)
{
ast_copy_string(buf, p-initreq.rlPart2, len);
-Original Message-
From: Steve Langstaff
Sent: 23 October 2006 09:58
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [asterisk-users] SIP_HEADER function; what names
are available?
Looking at the source code for Asterisk 1.2.7.1 (just what
I've got handy), it appears that the SIP_HEADER() function
just parses the SIP INVITE for whatever SIP *header* you specify - so:
a) there's no list of headers you can check for - it depends
on the user agent generating the request and
b) the request URI is not a SIP header, so you can't get to
it using a stock SIP_HEADER() function.
However, I suppose that there is nothing stopping you from
hacking the source for your Asterisk installation to provide
access to the URI... In chan_sip.c:func_header_read() you
could do something like:
static char *func_header_read(struct ast_channel *chan, char
*cmd, char *data, char *buf, size_t len) { snip/
content = get_header(p-initreq, data);
if (ast_strlen_zero(content)) {
new
/* look for an experimental pseudo-header that
allows us access to the request URI */
/* but note that this is not a real header name! */
if (strcmp(data,
x-Asterisk-Request-URI-pseudo-header)==0)
{
ast_copy_string(buf, p-initreq.rlPart2, len);
ast_mutex_unlock(chan-lock);
return buf;
}
/new
ast_mutex_unlock(chan-lock);
return NULL;
}
snip/
}
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf
Of Ricardo
Carvalho
Sent: 20 October 2006 17:51
To: kjcsb; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] SIP_HEADER function; what names are
available?
Any news on this thread? I also need to know the way to get
the R-URI
from sip INVITE messages received by Asterisk, through
${SIP_HEADER()}.
Thanks in advance,
Ricardo.
kjcsb wrote:
I have read the wiki about the SIP_HEADER function
(http://www.voip-
info.org/wiki/index.php?page=Asterisk+func+sip_header).
Where can I
get a list of the names that are available to be used with the
function e.g. TO is one name as in ${SIP_HEADER(TO)}.
What are the
others?
I would guess that you can check the RFC. Easier is to
turn on SIP
debug and see the INVITE packet yourself and check the
headers that
you have with your equipment.
/Olle
Thanks but I don't know how to get the actual INVITE details (the
request URI?). For example I want to get
sip:[EMAIL PROTECTED]
SIP/2.0 from the following dialogue:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Record-Route: sip:147.202.nn.nnn;ftag=bf7eced18eb7271b;lr=on
Via: SIP/2.0/UDP 147.202.nn.nnn;branch=z9hG4bKe49c.21b320a3.0
Via: SIP/2.0/UDP 60.234.nnn.nnn;branch=z9hG4bK76bf3dec8d45b972
From: User
sip:[EMAIL PROTECTED];tag=bf7eced18eb7271b
To: sip:[EMAIL PROTECTED]
etc
I can get Record-Route, Via, From, To etc but don't know
how to get
the bit after the INVITE. Interestingly only the first Via
is returned
by ${SIP_HEADER(VIA)}.
I've tried R-URI, RURI, URI, ALL, *, blank.
Any advice appreciated.
Cameron
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RE: [asterisk-users] Why does it take at least 4 flipping days before asterisk tries to resolve a provider?
On Mon, 23 Oct 2006, Andreas Sikkema wrote: Remco, Asterisk starts before the internet connection is up and dns is working. knip And then people say nightly asterisk restarts are not a good idea Why is your asterisk startup script running before networking has been setup? Asterisk has the same networking dependencies as apache, so I start it around the same time using the same priority as apache and as far as I know networking should work at that time or not at all, not somewhere in between. It is not, asterisk is correctly started after networking services, however it seems that when the box is booting the dns is replying just a split second too late for the taste of asterisk and it seems that asterisk then marks the provider as unavailable. * should never wait that long, the 'load' on the box to resolve maybe a handful of domains is nothing, even if you would be running a Pentium 1 box, and this should not be any reason not to try again every few minutes or so. pebkac? If your view is broad enough all computer / it related trouble could be traced back to that :) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] spandsp and freebsd
Hi,I have problem installing spandsp-0.0.3pre24 on FreeBSD 6.1. I get error: configure: error: Can't build without libtiff . But I have installed tiff from port tiff-3.8.2. I understand that the problem is about libtiff, and spandsp can't find these libs. So how to fix the problem? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] astdb error, please help
I checked the file permissions. They are proper. There doesnot seem to be a visible error. No change has been done in any conf files for the past 4 months. With warm regards. Vivek J. Joshi. [EMAIL PROTECTED] Trikon electronics Pvt. Ltd. All science is either physics or stamp collecting. -- Ernest Rutherford ram wrote: check database On 10/23/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hello friends, I am getting this error:- Oct 23 15:47:22 WARNING[2124]: db.c:171 ast_db_put: Unable to put value ' 192.168.1.12:5060:300:15553695861:sip:[EMAIL PROTECTED]:5060' for key '23' in family 'SIP/Registry I have no idea what it means. Please tell me what could be the problem. With warm regards. Vivek J. Joshi. [EMAIL PROTECTED] Trikon electronics Pvt. Ltd. All science is either physics or stamp collecting. -- Ernest Rutherford ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to deploy a PBX in such a condition ?
Soft phones or hard phones ? For softphones you can just use the PC's. If you go with hard phones you may want to get phones with QOS or build a seperate network for the phones. - Original Message - From: Bo Yang [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Monday, October 23, 2006 4:35 AM Subject: [asterisk-users] How to deploy a PBX in such a condition ? -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 My organization has a LAN now , and there almost a computer in each office for each employee . And in such a situation , what the most economic way to deploy a PBX with asterisk ? Is there good tutorials for me to learn how to do ? -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFFPCpm7tZp58UCwyMRAtGvAJ9koQF0Gzg8wxM8K+U01lwBOyenbACfcWu5 jZ68myehj2wrbzYosClWVCg= =Zg/t -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Zap Channel and VM problem
Title: Zap Channel and VM problem Hello, I am experiencing problems with a ZAP channel not being released after a voicemail has been left. I have 2 analogue extns from an Alcatel PBX wired into 2 FXO ports on my * box. If I make a call from an Alcatel extn and route it to a SIP extension on my * everything works as expected, at the end of the call the ZAP channel is released. As soon as I activate VM on the SIP extension VM can be left but the inbound ZAP channel is not released until 4 and a half minutes later (always). Users are simply putting the alcatel extn handset down to end the call (no key pad presses, no DTMF sent as * vm prompt suggests) During this time the SIP extension cannot receive any other calls from the Alcatel (the ZAP channel is busy). Am I right in thinking that the ZAP channel is timing out and releasing itself on a predetermined silence period (hence 4 and half minutes every time) If this is the case where do I change the ZAP (or is it VM) silence detect setting Regards Andy Green IT Manager GB eye Ltd 1 Russell St Kelham Island Sheffield S3 8RW Tel: 0114 252 1611 Fax: 0114 272 9599 mailto:[EMAIL PROTECTED] http://www.gbeye.com Coming Soon: WWE, Eregon (Fox Films), Torchwood (BBC), Iron Maiden, Terminator, Bill andTed, Wedding Crashers, Muse,Reservoir Dogs (Prints & Canvas). Check out our latest releases: http://www.gbeye.com/newsite/releases/ Or browse our whole catalogue: http://www.gbeye.com/newsite/catalogue/ This e-mail is intended for the addressee(s) named above and any other use is prohibited. It may contain confidential information.If you received this e-mail in error please contact the sender by return e-mail. GB eye Ltd does not accept legal responsibility for the contents of this message if it has reached you via the Internet. Any opinions expressed are those of the author and are not necessarily endorsed by GB eye Ltd. Recipients are advised to apply their own virus checks to this message and all incoming email on delivery. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] spandsp and freebsd
On 10/23/06, Giedrius Augys [EMAIL PROTECTED] wrote: Hi, I have problem installing spandsp-0.0.3pre24 on FreeBSD 6.1. I get error: configure: error: Can't build without libtiff . But I have installed tiff from port tiff-3.8.2. I understand that the problem is about libtiff, and spandsp can't find these libs. So how to fix the problem? Thanks Is it possible that the Makefile looks for headers in /usr/include, and ports has included them in /usr/local/include? If so, just mangle the Makefile to suit. Cheers, Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zap Channel and VM problem
On Mon, Oct 23, 2006 at 12:52:49PM +0100, Andy Green wrote: Hello, I am experiencing problems with a ZAP channel not being released after a voicemail has been left. I have 2 analogue extns from an Alcatel PBX wired into 2 FXO ports on my * box. If I make a call from an Alcatel extn and route it to a SIP extension on my * everything works as expected, at the end of the call the ZAP channel is released. As soon as I activate VM on the SIP extension VM can be left but the inbound ZAP channel is not released until 4 and a half minutes later (always). Who is hanging up the call? Asterisk or the Alcatel PBX? If the Alcatel PBX: how does it notify the hangup? Users are simply putting the alcatel extn handset down to end the call (no key pad presses, no DTMF sent as * vm prompt suggests) During this time the SIP extension cannot receive any other calls from the Alcatel (the ZAP channel is busy). Am I right in thinking that the ZAP channel is timing out and releasing itself on a predetermined silence period (hence 4 and half minutes every time) If this is the case where do I change the ZAP (or is it VM) silence detect setting -- Tzafrir Cohen iax:[EMAIL PROTECTED]/tzafrir icq#16849755 mailto:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] (no subject)
Hi All I would greatly appreciate some advice or some direction as to where to go next. I have a provider passing me incoming calls via my Session Border Controller. I am able to pass them calls fine but coming in fails with a 407 Authentication Fail error. In my sip.conf I have an entry for the provider but am not asking for a user/pass so I would expect the calls to come in and then pass to the context specified in extensions.conf: [iplcr-gw] type=peer host=xx.xx.xx.xx nat=no dtmfmode=inband context=from-iplcr insecure=invite canreinvite=yes disallow=all allow=ulaw,alaw I have tried different insecure= methods but am still getting the same error. Does anyone know what else could be causing the error or suggest some other things I should try? Many Thanks Scott ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Real Time and Asterisk
I'm investigating in deploying an asterisk solution. After some experiments, I noticed that asterisk is quite subect to unreproducible troubles and quality losses with variations of server load, and also a lot of troubles that could be reconducted to timing problems, expecially with faxes. I was wondering, would asterisk take benefits from being deployed on a real-time enabled linux? I tried making some researches on the net, but usually searching realtime and asterisk mean finding informations about the realtime database configuration of asterisk... Before investing more effort in investigating the not simple world of real time Oses, i wanted to know if somebody have some suggestions about that... Thanks, Francesco ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problems with chan-capi and Eicon Diva 4BRI
Hi! This weekend we had a problem with our Asterisk Box which ran flawlessly for nearly 4 weeks. The Asterisk server sits between the PSTN and a Siemens PBX and bridges 2 BRI lines. No calls, not incoming, not outgoing. The admin rebooted the Dell Box and then everything worked fine again. Now, I'm analyzing log files to find the cause. During the Asterisk outage the logfiles only show incoming (PSTN-Asterisk-PBX) calls, no outgoing. Thus I suspect that the Asterisk--PBX link was broken. In the Asterisk message file I only see Recovery on timer expiry errors, like below: Oct 20 17:18:18 VERBOSE[19772] logger.c: == ISDN2#02: Incoming call '347x' - '32xx' Oct 20 17:18:18 VERBOSE[19772] logger.c: -- ISDN2#02: Updated channel name: CAPI/ISDN2/32xx-8ab6 Oct 20 17:18:18 VERBOSE[2663] logger.c: -- Executing Dial(CAPI/ISDN2/32xx-8ab6, CAPI/g2//b|90) in new stack Oct 20 17:18:18 VERBOSE[2663] logger.c: -- Called g2//b Oct 20 17:18:19 VERBOSE[19772] logger.c: -- ISDN2#02: Updated channel name: CAPI/ISDN2/32xx11-8ab8 Oct 20 17:18:19 VERBOSE[2663] logger.c: -- ISDN4#02: Updated channel name: CAPI/ISDN4/1-8ab9 Oct 20 17:18:19 VERBOSE[2663] logger.c: -- ISDN4#02: Updated channel name: CAPI/ISDN4/11-8aba Oct 20 17:18:26 VERBOSE[19772] logger.c: ISDN4#02: CAPI INFO 0x34e6: Recovery on timer expiry Oct 20 17:18:26 VERBOSE[2663] logger.c: == ISDN4#02: CAPI Hangingup for PLCI=0x104 in state 4 Oct 20 17:18:26 VERBOSE[2663] logger.c: == Everyone is busy/congested at this time (1:0/0/1) Oct 20 17:18:26 VERBOSE[2663] logger.c: -- Executing Hangup(CAPI/ISDN2/32xx11-8ab8, ) in new stack Oct 20 17:18:26 VERBOSE[2663] logger.c: == Spawn extension (frompstn, 32xx, 2) exited non-zero on 'CAPI/ISDN2/32xx11-8ab8' Oct 20 17:18:26 VERBOSE[2663] logger.c: == ISDN2#02: CAPI Hangingup for PLCI=0x202 in state 7 Oct 20 17:18:26 VERBOSE[19772] logger.c: ISDN2#02: CAPI INFO 0x34e6: Recovery on timer expiry What does the timer expiry exactly mean? Was it a Layer2 or Layer 3 problem? How can I find out more or how can I activate more BRI debugging for the case it happens again? Are there any known problems? We are using: Asterisk 1.2.12.1 chan_capi-0.7.0 divas4linux-melware-3.0.3-106.650-1 Diva Server 4BRI-8M 2.0 PCI Thanks Klaus ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: checking 'voicemail externally - doesn't work
Joseph wrote:
Though what option am I suppose to pass it.
The process seems to me correct, when I get-in to disa-access I have
access to voicemail extension 1000 (otherwise it wouldn't let me dial
ext. 1000; when I dial it it asking me for mailbox number and password,
except that password is not recognized; even tough I see it from the
command line that the correct password 123 was entered. So I don't
understand why isn't it accepting it?
[voicemail]
exten = 1000,1,NoCDR()
exten = 1000,2,Answer()
exten = 1000,3,VoicemailMain(${CALLERIDNUM})
Looks at the console log again. You should be seeing
VoicemailMain(1235551212) or whatever telephone number you are calling
from. Is the telephone number you are calling from the same as the
mailbox name in voicemail.conf?
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RE: [asterisk-users] Can't change Zaptel driver from FCC mode
Hi Tzafrir, If I do a cat of /sys/module/wctdm/opermode I get: FCC I thought I could change it here and then do a ztcfg or a genzaptelconf but it just overwrites it with FCC again. How do I change it? Neil safeharbour IT Ltd Your IT Department tel: 0845 644 3607 fax: 0845 867 2891 mob: 07812 114784 voip: [EMAIL PROTECTED] email: [EMAIL PROTECTED] web: www.safeharbourit.co.uk The information in this e-mail is confidential and may be legally privileged. It is intended solely for the addressee. Access to this e-mail by anyone else is unauthorised. If you are not the intended recipient, any disclosure, copying, distribution or any action taken or omitted to be taken in reliance on it, is prohibited and may be unlawful. When addressed to our clients, any opinions or advice contained in this e-mail are subject to the terms and conditions expressed in any applicable governing terms of business. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: 23 October 2006 10:41 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Can't change Zaptel driver from FCC mode On Mon, Oct 23, 2006 at 09:34:10AM +0100, Neil Tancock wrote: Hi, I'm using Asterisk with a Digium TDM10B FXO card and it's driving me nuts. I'm based in the UK and have echo problems and need to switch the driver from FCC mode to UK mode. I've tried modprobe zaptel and modprobe wctdm opermode=UK and the ztcfg. I get no error messages but when I reboot it still comes up as FCC mode? What is the actual parameter? Any chance it is set elsewhere or that the module was already loaded? To check the current value: cat /sys/modules/wctdm/parameters/opermode Also verify you don't use the parameter _opermode. -- Tzafrir Cohen iax:[EMAIL PROTECTED]/tzafrir icq#16849755 mailto:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] using asterisk to do remote control functions
Ok didn't know TrixBox had remote, control support ... If anyone has please tell .. Thanks greg On 10/20/06, Matthew Rubenstein [EMAIL PROTECTED] wrote: Has anyone used the TrixBox/AAH builtin facility xPL for facility (including home/office/industrial) automation? On Fri, 2006-10-20 at 05:17 -0700, [EMAIL PROTECTED] wrote: Date: Fri, 20 Oct 2006 11:28:51 +0200 From: Gregory Machin [EMAIL PROTECTED] Subject: [asterisk-users] using asterisk to do remote control functions To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=UTF-8; format=flowed Hi Im very green to asterisk, and I have been asked if asterisk can be used to do remote control, like opening gates etc, say when the user dials a predefined number ... And what hardware is required ... Many Thanks -- Gregory Machin -- (C) Matthew Rubenstein -- Gregory Machin [EMAIL PROTECTED] www.linuxpro.co.za ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] spandsp and freebsd
2006/10/23, Steve Davies [EMAIL PROTECTED]: On 10/23/06, Giedrius Augys [EMAIL PROTECTED] wrote: Hi, I have problem installing spandsp-0.0.3pre24 on FreeBSD 6.1.I get error: configure: error:Can't build without libtiff . But I have installed tiff from port tiff-3.8.2. I understand that the problem is about libtiff, and spandsp can't find these libs. So how to fix the problem? ThanksIs it possible that the Makefile looks for headers in /usr/include, and ports has included them in /usr/local/include? If so, just manglethe Makefile to suit.Cheers,Steve___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Is it possible during configure to show where to find libtiff librarie? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (no subject)
You might want to repost it with a subject or you miss a lot of people seeing or opening it up. -- Original message -- From: "Scott Pinhorne" [EMAIL PROTECTED] Hi All I would greatly appreciate some advice or some direction as to where to go next. I have a provider passing me incoming calls via my Session Border Controller. I am able to pass them calls fine but coming in fails with a 407 Authentication Fail error. In my sip.conf I have an entry for the provider but am not asking for a user/pass so I would expect the calls to come in and then pass to the context specified in extensions.conf: [iplcr-gw] type=peer host=xx.xx.xx.xx nat=no dtmfmode=inband context=from-iplcr insecure=invite canreinvite=yes disallow=all allow=ulaw,alaw I have tried different insecure= methods but am still getting the same error. Does anyone know what else could be causing the error or suggest some other things I should try? Many Thanks Scott ---BeginMessage--- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ---End Message--- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problems with chan-capi and Eicon Diva 4BRI
Hi Klaus. I'm not sure about the timer expiry meaning, but you could use the xlog command (usually found in /usr/lib/eicon/divas) Just run it as root indicating which span (1..4) you want to trace: ./xlog -c 2 that shoud show you layer 1 layer 2 dump Alberto. Klaus Darilion ha scritto: Hi! This weekend we had a problem with our Asterisk Box which ran flawlessly for nearly 4 weeks. The Asterisk server sits between the PSTN and a Siemens PBX and bridges 2 BRI lines. No calls, not incoming, not outgoing. The admin rebooted the Dell Box and then everything worked fine again. Now, I'm analyzing log files to find the cause. During the Asterisk outage the logfiles only show incoming (PSTN-Asterisk-PBX) calls, no outgoing. Thus I suspect that the Asterisk--PBX link was broken. In the Asterisk message file I only see Recovery on timer expiry errors, like below: Oct 20 17:18:18 VERBOSE[19772] logger.c: == ISDN2#02: Incoming call '347x' - '32xx' Oct 20 17:18:18 VERBOSE[19772] logger.c: -- ISDN2#02: Updated channel name: CAPI/ISDN2/32xx-8ab6 Oct 20 17:18:18 VERBOSE[2663] logger.c: -- Executing Dial(CAPI/ISDN2/32xx-8ab6, CAPI/g2//b|90) in new stack Oct 20 17:18:18 VERBOSE[2663] logger.c: -- Called g2//b Oct 20 17:18:19 VERBOSE[19772] logger.c: -- ISDN2#02: Updated channel name: CAPI/ISDN2/32xx11-8ab8 Oct 20 17:18:19 VERBOSE[2663] logger.c: -- ISDN4#02: Updated channel name: CAPI/ISDN4/1-8ab9 Oct 20 17:18:19 VERBOSE[2663] logger.c: -- ISDN4#02: Updated channel name: CAPI/ISDN4/11-8aba Oct 20 17:18:26 VERBOSE[19772] logger.c: ISDN4#02: CAPI INFO 0x34e6: Recovery on timer expiry Oct 20 17:18:26 VERBOSE[2663] logger.c: == ISDN4#02: CAPI Hangingup for PLCI=0x104 in state 4 Oct 20 17:18:26 VERBOSE[2663] logger.c: == Everyone is busy/congested at this time (1:0/0/1) Oct 20 17:18:26 VERBOSE[2663] logger.c: -- Executing Hangup(CAPI/ISDN2/32xx11-8ab8, ) in new stack Oct 20 17:18:26 VERBOSE[2663] logger.c: == Spawn extension (frompstn, 32xx, 2) exited non-zero on 'CAPI/ISDN2/32xx11-8ab8' Oct 20 17:18:26 VERBOSE[2663] logger.c: == ISDN2#02: CAPI Hangingup for PLCI=0x202 in state 7 Oct 20 17:18:26 VERBOSE[19772] logger.c: ISDN2#02: CAPI INFO 0x34e6: Recovery on timer expiry What does the timer expiry exactly mean? Was it a Layer2 or Layer 3 problem? How can I find out more or how can I activate more BRI debugging for the case it happens again? Are there any known problems? We are using: Asterisk 1.2.12.1 chan_capi-0.7.0 divas4linux-melware-3.0.3-106.650-1 Diva Server 4BRI-8M 2.0 PCI Thanks Klaus ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- -- Alberto Pastore B-Press Srl - Gruppo MSoft P.IVA 01697420030 P.le Lombardia, 4 - 28100 Novara - Italy Tel. 0321-499508 Fax 0321-492974 http://www.msoft.it ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoicePulse Connect 4 Channel Limit?
Good prices means (exactly) reasonable prices. I´m a newbie, so I´m asking for good experiences... Thanks in advance... R.R. Libera Lacy Moore - Aspendora escribió: So, What´s your recommendation for a production environment? I was looking for good prices, good voice quality for USA Origination and I´d like to hear about good experiences PSTN. Just can't beat the quality :-) Wait, you said good prices. Sorry. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Why does it take at least 4 flipping days before asterisk tries to resolve a provider?
Remco Barendse wrote: It is not, asterisk is correctly started after networking services, however it seems that when the box is booting the dns is replying just a split second too late for the taste of asterisk and it seems that asterisk then marks the provider as unavailable. * should never wait that long, the 'load' on the box to resolve maybe a handful of domains is nothing, even if you would be running a Pentium 1 box, and this should not be any reason not to try again every few minutes or so. I was under the impression that it only contacted the hosts if it was registering with them, then it would wait until the value passed in registertimeout (sip.conf) and retry again after that time (if it failed). In practise here (and at work) if the machine has problems contacting registration hosts, anyx sip clients connected to the server (even locally) will not register and any fixed sip peer - user pairs that don't require registration will also not work until it can contact the host again. I doubt this is what's supposed to happen, but that's what happens with me (1.2.12.1 and 1.4b2). ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CID Issues
Hello, I've posted this at the trixbox and freepbx forums and haven't been able to get an answer. I thought perhaps the guru's here might be able to help me out :) I'm having some issues with setting caller IDs. There are 2 problems that I would like to solve. 1. I have a DID pointing to a ring group. The only 'extension' in that ring group is an external number (cell phone). So essentially the DID fwds to a cell phone. The problem is that the CID that shows up on the Cell phone is the number that's set on the outgoing trunk the the CALLERS #. Is there a simple way to override this? Or better yet, is there a prefered method for forwarding calls out with freePBX? 2. I have sevaral trixbox installs connected through DUNDI. The DUNDI works very well.. I can call local extensions from every PBX. The PBX's are connected via an IAX trunk. In freePBX I've created a custom trunk that accepts a 4 digit extension and puts the call into a 'trydundi' context. The problem I'm having is that whenever someone calls from an extension at one location to an extension at another location the CallerID that shows up at the other location is the one set either in #1 The custom trunk, or #2 in the 'Outbound CID' field in the users screen. What I WANT this to be set to is the Name of the extension ie. just like local calls are. Is there a way to do this painlessly. Is it possible to hook dundi into a different context so that it would think all calls are local.. I'm kinda guessing here. Sorry about the length of these descriptions and thanks for any advice! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] spandsp and freebsd
On Mon, Oct 23, 2006 at 02:32:55PM +0300, Giedrius Augys wrote: Hi, I have problem installing spandsp-0.0.3pre24 on FreeBSD 6.1. I get error: configure: error: Can't build without libtiff . But I have installed tiff from port tiff-3.8.2. I understand that the problem is about libtiff, and spandsp can't find these libs. So how to fix the problem? Thanks Try: env CPPFLAGS=-I/usr/local/include LDFLAGS=-L/usr/local/lib ./configure ...etc ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] rxfax problem
Grab the fax2mail script from www.generationd.com and set it to convert the tiff to pdf before sending. Works great. MD -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Davies Sent: Monday, October 23, 2006 4:38 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] rxfax problem On 10/20/06, Mohammad Shokuie [EMAIL PROTECTED] wrote: Anyways, let me take the most benefit as im sure you'd read this post, i have problem with the size of received page which is shrinked, can u give me a hint about this problem too :) This is probably the problem of the application that you use to view the TIFF file. FAX machines generate TIFF files with different horizontal and vertical resolution, and a lot of lazy programs do not check this correctly. I find that a quick 'tiff2pdf' conversion fixes things up very nicely :) Steve D ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Primary D-Channel channal numbers....
Greetings, All! Help to find the reason, constantly writes: == Primary D-Channel on span 2 up --- Log: == Primary D-Channel on span 2 up == Primary D-Channel on span 2 down Oct 23 10:26:25 WARNING[9515]: chan_zap.c:2287 pri_find_dchan: No D-channels available! Using Primary channel 47 as D-channel anyway! == Primary D-Channel on span 2 up == Primary D-Channel on span 2 up == Primary D-Channel on span 2 up == Primary D-Channel on span 2 up . Accordingly on it span nothing goes What can be to it the reason? On span 2 such messages have started to go after the operator at itself has started some utility for testing... Not clearest for me that I setup span 3 also as well as span 2 and all on span 3 has earned, and on 2-nd and has not risen... Your offers?! P.S.: Prompt as to register channels in *, that to each channel there corresponded number??? begin:vcard fn:Eugeniy Khvastunov n:Khvastunov;Eugeniy org:Digma;IT adr:;;;Kharkov;Kh;;Ukraine email;internet:[EMAIL PROTECTED] title:System Administrator tel;work:+380675745646 tel;cell:+380504063116 version:2.1 end:vcard ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 7960/SIP MWI Question
The 7960's have an envelope that appears in the display next to a line which has voicemail. Also, the MWI light is a logical OR of all the defined lines. Is there a way to tell the phone NOT to display the MWI for certain lines but retain the envelope for all? If you get enough VM on busy lines then the light tends to lose meaning and you may as well have it on all the time! I'm currently on POS3-06-3-00 dbc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CID Issues
On 10/23/06, mail-lists [EMAIL PROTECTED] wrote: Hello,I've posted this at the trixbox and freepbx forums and haven't been ableto get an answer. I thought perhaps the guru's here might be able tohelp me out :)I'm having some issues with setting caller IDs. There are 2 problems that I would like to solve.1. I have a DID pointing to a ring group. The only 'extension' in thatring group is an external number (cell phone). So essentially the DIDfwds to a cell phone. The problem is that the CID that shows up on the Cell phone is the number that's set on the outgoing trunk the theCALLERS #. Is there a simple way to override this? Or better yet, isthere a prefered method for forwarding calls out with freePBX? Are you allowed to set your own CallerID on outbound calls from your provider? 2. I have sevaral trixbox installs connected through DUNDI. The DUNDIworks very well.. I can call local extensions from every PBX. The PBX'sare connected via an IAX trunk. In freePBX I've created a custom trunk that accepts a 4 digit extension and puts the call into a 'trydundi'context. The problem I'm having is that whenever someone calls from anextension at one location to an extension at another location theCallerID that shows up at the other location is the one set either in #1 The custom trunk, or #2 in the 'Outbound CID' field in the users screen.What I WANT this to be set to is the Name of the extension ie. just likelocal calls are. Is there a way to do this painlessly. Is it possible to hook dundi into a different context so that it would think all calls arelocal.. I'm kinda guessing here.Sorry about the length of these descriptions and thanks for any advice!___ --Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Tom VileBaldwin Technology Solutions, IncConsulting - Web Design - VoIP Telephony www.baldwintechsolutions.comPhone: 518-631-2855 x205Fax: 518-631-2856 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] chan_h323.so Asterisk Beta compilation
I have had some interesting compiling results with the latest beta release of Asterisk. With reference to this channel After running the make opt in the H323 directory, and the make install in the Asterisk directory, there is still no chan_h323.so file Created.. Are there any other args or commands that need to be set to get this to work? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Primary D-Channel channal numbers....
Hello, I had the same trouble, it was the telco operator that had an equipment in fault... It was unable to get the RNIS level 2 communication up... You should issue: pri intense debug span 2 and see what happens to the line... If you see SABME msg going in and out in loop, it is what i had... Eugeniy Khvastunov a écrit : Greetings, All! Help to find the reason, constantly writes: == Primary D-Channel on span 2 up --- Log: == Primary D-Channel on span 2 up == Primary D-Channel on span 2 down Oct 23 10:26:25 WARNING[9515]: chan_zap.c:2287 pri_find_dchan: No D-channels available! Using Primary channel 47 as D-channel anyway! == Primary D-Channel on span 2 up == Primary D-Channel on span 2 up == Primary D-Channel on span 2 up == Primary D-Channel on span 2 up . Accordingly on it span nothing goes What can be to it the reason? On span 2 such messages have started to go after the operator at itself has started some utility for testing... Not clearest for me that I setup span 3 also as well as span 2 and all on span 3 has earned, and on 2-nd and has not risen... Your offers?! P.S.: Prompt as to register channels in *, that to each channel there corresponded number??? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can't change Zaptel driver from FCC mode
On Mon, Oct 23, 2006 at 01:46:53PM +0100, Neil Tancock wrote: Hi Tzafrir, If I do a cat of /sys/module/wctdm/opermode I get: FCC I thought I could change it here No point in changing it there. It is used when the module is initialized (and should be a read-only parameter, hint-hint) and then do a ztcfg ztcfg is irrelevant here: it only operates after the module is loaded. or a genzaptelconf Again, not exactly relevant but it just overwrites it with FCC again. Have you unloaded the kernel module wctdm? Again: 'lsmod | grep wctdm' shows if it is loaded. grep wctdm /etc/modprobe.conf -- Tzafrir Cohen iax:[EMAIL PROTECTED]/tzafrir icq#16849755 mailto:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CID Issues
- Original Message - From: mail-lists [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, October 23, 2006 3:39 PM Subject: [asterisk-users] CID Issues Hello, I've posted this at the trixbox and freepbx forums and haven't been able to get an answer. I thought perhaps the guru's here might be able to help me out :) I'm having some issues with setting caller IDs. There are 2 problems that I would like to solve. 1. I have a DID pointing to a ring group. The only 'extension' in that ring group is an external number (cell phone). So essentially the DID fwds to a cell phone. The problem is that the CID that shows up on the Cell phone is the number that's set on the outgoing trunk the the CALLERS #. Is there a simple way to override this? Or better yet, is there a prefered method for forwarding calls out with freePBX? I dont mean to be harsh but learn the real asterisk. trixbox is a crutch for asterisk. I have not used trixbox enough to know how they generate the dialplan files to tell you how to mod them. One thing you can check is to see if your VOIP provider allows you to set the caller ID at all. I know some providers wont even let you change it. 2. I have sevaral trixbox installs connected through DUNDI. The DUNDI works very well.. I can call local extensions from every PBX. The PBX's are connected via an IAX trunk. In freePBX I've created a custom trunk that accepts a 4 digit extension and puts the call into a 'trydundi' context. The problem I'm having is that whenever someone calls from an extension at one location to an extension at another location the CallerID that shows up at the other location is the one set either in #1 The custom trunk, or #2 in the 'Outbound CID' field in the users screen. What I WANT this to be set to is the Name of the extension ie. just like local calls are. Is there a way to do this painlessly. Is it possible to hook dundi into a different context so that it would think all calls are local.. I'm kinda guessing here. Sorry about the length of these descriptions and thanks for any advice! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] chan_h323.so Asterisk Beta compilation
On Mon, Oct 23, 2006 at 04:09:43PM +0200, Patrick wrote: I have had some interesting compiling results with the latest beta release of Asterisk.. With reference to this channel. After running the make opt in the H323 directory, and the make install in the Asterisk directory, there is still no chan_h323.so file Created.. Are there any other args or commands that need to be set to get this to work? For the module to be built you need autoconf to detect your version of openh323 (and pwlib), and to have that module selected. When you run 'menuselect', and enter the channels section, do you see the module chan_h323: selected, unselected, or XXX-ed out? -- Tzafrir Cohen iax:[EMAIL PROTECTED]/tzafrir icq#16849755 mailto:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Can't change Zaptel driver from FCC mode
Ok, I've somehow resolved this. I've added the line options wctdm opermode=UK to modprobe.conf, rebuilt zaptel and now when it boots I get UK mode instead of FCC mode and, hey presto, less echo! Many thanks Tzafrir! Neil safeharbour IT Ltd Your IT Department tel: 0845 644 3607 fax: 0845 867 2891 mob: 07812 114784 voip: [EMAIL PROTECTED] email: [EMAIL PROTECTED] web: www.safeharbourit.co.uk The information in this e-mail is confidential and may be legally privileged. It is intended solely for the addressee. Access to this e-mail by anyone else is unauthorised. If you are not the intended recipient, any disclosure, copying, distribution or any action taken or omitted to be taken in reliance on it, is prohibited and may be unlawful. When addressed to our clients, any opinions or advice contained in this e-mail are subject to the terms and conditions expressed in any applicable governing terms of business. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: 23 October 2006 15:17 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Can't change Zaptel driver from FCC mode On Mon, Oct 23, 2006 at 01:46:53PM +0100, Neil Tancock wrote: Hi Tzafrir, If I do a cat of /sys/module/wctdm/opermode I get: FCC I thought I could change it here No point in changing it there. It is used when the module is initialized (and should be a read-only parameter, hint-hint) and then do a ztcfg ztcfg is irrelevant here: it only operates after the module is loaded. or a genzaptelconf Again, not exactly relevant but it just overwrites it with FCC again. Have you unloaded the kernel module wctdm? Again: 'lsmod | grep wctdm' shows if it is loaded. grep wctdm /etc/modprobe.conf -- Tzafrir Cohen iax:[EMAIL PROTECTED]/tzafrir icq#16849755 mailto:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] chan_h323.so Asterisk Beta compilation
When I have a look and the menuselect.makeopts file.. MENUSELECT_CHANNELS= chan_gtalk chan_h323 ... is there, and also during the ./configure, all the various pwlib and openh323 version checks seem valid.. but still not sure where you enable the channel to be built... According to the H323 README, it just says make opt, then go to the asterisk directory, and make install, but that still has no effect because again the actual chan_h323.so file is not built... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: 23 October 2006 04:29 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] chan_h323.so Asterisk Beta compilation On Mon, Oct 23, 2006 at 04:09:43PM +0200, Patrick wrote: I have had some interesting compiling results with the latest beta release of Asterisk.. With reference to this channel. After running the make opt in the H323 directory, and the make install in the Asterisk directory, there is still no chan_h323.so file Created.. Are there any other args or commands that need to be set to get this to work? For the module to be built you need autoconf to detect your version of openh323 (and pwlib), and to have that module selected. When you run 'menuselect', and enter the channels section, do you see the module chan_h323: selected, unselected, or XXX-ed out? -- Tzafrir Cohen iax:[EMAIL PROTECTED]/tzafrir icq#16849755 mailto:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to busy out PRI channels?
Is there any way under software control (CLI, Manager, etc.) to busy out one or more PRI channels, for testing purposes, without actually having to make real calls on them? It should have the following effects: a) Outgoing calls will not try to use the busied out channels, when using a group specifier. An attempt explicitly to dial via a busied channel would fail with CHANUNAVAIL. b) The remote switch would recognise the channels as busy and would hunt for a non-busy channel when attempting to place a call to the system. The intention is to simulate behaviour as a system approaches capacity, not in terms of loading but of routing, etc. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Macro 'exited non-zero'
Can someone tell me if this indicates a problem? What does it mean when a macro exits != 0 ? Spawn extension (macro-syst_FindAppServer, s, 5) exited non-zero on 'SIP/xxx.yyy.142.186-b7515f98' in macro 'syst_FindAppServer' Thanks, Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Why does it take at least 4 flipping days before asterisk tries to resolve a provider?
Thomas Kenyon wrote: Remco Barendse wrote: It is not, asterisk is correctly started after networking services, however it seems that when the box is booting the dns is replying just a split second too late for the taste of asterisk and it seems that asterisk then marks the provider as unavailable. * should never wait that long, the 'load' on the box to resolve maybe a handful of domains is nothing, even if you would be running a Pentium 1 box, and this should not be any reason not to try again every few minutes or so. I was under the impression that it only contacted the hosts if it was registering with them, then it would wait until the value passed in registertimeout (sip.conf) and retry again after that time (if it failed). In practise here (and at work) if the machine has problems contacting registration hosts, anyx sip clients connected to the server (even locally) will not register and any fixed sip peer - user pairs that don't require registration will also not work until it can contact the host again. I doubt this is what's supposed to happen, but that's what happens with me (1.2.12.1 and 1.4b2). This is a known issue. See the mailing list archives. Make sure asterisk does not try to resolve using DNS. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 2621 NM-HDV VWIC-1MFT1
Thanks..
Did I misread the posts? These look like the VWIC-1MFT-T1 is connecting to
the PSTN and not connecting to a PRI card on an Asterisk box..
We are looking to do the following..
Asterisk PRI card - VWIC-1MFT-T1 - SIP -
Thanks
David
- Original Message -
From: Tijl Van den Broeck [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, October 18, 2006 3:58 AM
Subject: Re: [asterisk-users] Cisco 2621 NM-HDV VWIC-1MFT1
http://www.voip-info.org/wiki/index.php?page=Asterisk+cisco+FXO is a
good read for that.
I've got a couple of 2600's configured this way, and all seems to work
just fine. One little detail I came across was one-way-audio..
strangely enough that was fixed if I used
Dial(SIP/${EXTEN:[EMAIL PROTECTED],40,to) .. the o Dial-option fixed
it in my dialplan, both for outgoing and incoming calls.
SIP calls from the 2600 arrive in your asterisk in the form
[EMAIL PROTECTED], my approach was to let it use default context, then
match the numbers there with exten and send it off to the individual
contexts from there with Gosub().
Good luck :-)
On 10/18/06, David Edwards [EMAIL PROTECTED] wrote:
Steve,
I was just looking for a little info to get me started..
Thanks
David
- Original Message -
From: Steve Blair [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, October 17, 2006 19:24
Subject: Re: [asterisk-users] Cisco 2621 NM-HDV VWIC-1MFT1
David:
Do you have a specific problem with this card? If not and you are just
looking for general information you can try the following document.
-Steve
http://mit.edu/sip/sip.edu/ciscoGW.html
David Edwards wrote:
Hi all,
We are trying to use a Cisco 2621 with NM-HDV VWIC-1MFT1 to connect
to a PBX via the PRI card. We want to use it as a gateway to forward
all calls to a hosted Asterisk server off-site via SIP.
Does any one have any suggestions on how to best approach this?
Thanks
David
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Re: [asterisk-users] CID Issues
mail-lists wrote: Hello, I've posted this at the trixbox and freepbx forums and haven't been able to get an answer. I thought perhaps the guru's here might be able to help me out :) I'm having some issues with setting caller IDs. There are 2 problems that I would like to solve. 1. I have a DID pointing to a ring group. The only 'extension' in that ring group is an external number (cell phone). So essentially the DID fwds to a cell phone. The problem is that the CID that shows up on the Cell phone is the number that's set on the outgoing trunk the the CALLERS #. Is there a simple way to override this? Or better yet, is there a prefered method for forwarding calls out with freePBX? I cannot help you with Trixbox, only Asterisk. Are you using PRI, Analog, or VoIP for your outgoing call? If PRI or VoIP AND the carrier permits it, you can manually set the Caller*ID before the Dial() line using SetCIDNum. You can also do a show application dial in the Asterisk CLI, pay special attention to the o option to Dial() ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 2621 NM-HDV VWIC-1MFT1
David Edwards wrote: Thanks.. Did I misread the posts? These look like the VWIC-1MFT-T1 is connecting to the PSTN and not connecting to a PRI card on an Asterisk box.. We are looking to do the following.. Asterisk PRI card - VWIC-1MFT-T1 - SIP - Why not have Asterisk connect directly to the remote SIP device/server. If the remote server is Asterisk, why not use IAX2? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk not detecting hangup
Hi, Im working with the following versions: -asterisk-1.2.12.1 -zaptel-1.2.9.1 And with the following card: 00:0d.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface Subsystem: Unknown device 8085:0003 Flags: bus master, medium devsel, latency 32, IRQ 201 I/O ports at c800 [size=256] Memory at fe00 (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 2 Identified as: *CLI zap show status Description Alarms IRQbpviol CRC4 Wildcard X101P Board 1 OK 0 0 0 And the following lines in zapata.conf(for spanish lines): answeronpolarityswitch=yes hanguponpolarityswitch=yes The problem is that although the calls work correctly the system is unable to detect a pstn hangup and it keeps running even when the other side is calling to another number(not an asterisk ones, asterisk line keeps busy) Any hint? Thanks for your time -- Arkaitz ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk and HMP
Hi all, Does Asterisk now support Intels HMP platforms? Does it support in 1.4 version ? Thanks. Greg ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 2621 NM-HDV VWIC-1MFT1
Sorry a typo.. Having one of those Mondays.. Non-IP PBX (PRI Interface) - VWIC-1MFT-T1 - SIP - Asterisk I am considering recommending/testing something like the Quintum Tenor products.. I like Cisco, but in this case it might not be the best option.. David - Original Message - From: Eric ManxPower Wieling [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, October 23, 2006 11:32 AM Subject: Re: [asterisk-users] Cisco 2621 NM-HDV VWIC-1MFT1 David Edwards wrote: Thanks.. Did I misread the posts? These look like the VWIC-1MFT-T1 is connecting to the PSTN and not connecting to a PRI card on an Asterisk box.. We are looking to do the following.. Asterisk PRI card - VWIC-1MFT-T1 - SIP - Why not have Asterisk connect directly to the remote SIP device/server. If the remote server is Asterisk, why not use IAX2? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: checking 'voicemail externally - doesn't work
On Mon, 2006-10-23 at 07:46 -0500, Eric ManxPower Wieling wrote:
Joseph wrote:
Though what option am I suppose to pass it.
The process seems to me correct, when I get-in to disa-access I have
access to voicemail extension 1000 (otherwise it wouldn't let me dial
ext. 1000; when I dial it it asking me for mailbox number and password,
except that password is not recognized; even tough I see it from the
command line that the correct password 123 was entered. So I don't
understand why isn't it accepting it?
[voicemail]
exten = 1000,1,NoCDR()
exten = 1000,2,Answer()
exten = 1000,3,VoicemailMain(${CALLERIDNUM})
Looks at the console log again. You should be seeing
VoicemailMain(1235551212) or whatever telephone number you are calling
from. Is the telephone number you are calling from the same as the
mailbox name in voicemail.conf?
At the console I see:
Executing VoiceMailMain(SIP/pstn-1270-0819a1f0, pstn1270) in new
stack
-- Playing 'vm-login' (language 'en')
-- Playing 'vm-password' (language 'en')
-- Incorrect password '123' for user 'tn12701' (context = default)
Maybe the problem is that the user tn12701 is not in the
voicemail.conf?
I don't think it is possible to assign two users to one mail box, is
it?. I have to make two entries, I think.
--
#Joseph
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Re: [asterisk-users] Re: checking 'voicemail externally - doesn't work
Joseph wrote:
On Mon, 2006-10-23 at 07:46 -0500, Eric ManxPower Wieling wrote:
Joseph wrote:
Though what option am I suppose to pass it.
The process seems to me correct, when I get-in to disa-access I have
access to voicemail extension 1000 (otherwise it wouldn't let me dial
ext. 1000; when I dial it it asking me for mailbox number and password,
except that password is not recognized; even tough I see it from the
command line that the correct password 123 was entered. So I don't
understand why isn't it accepting it?
[voicemail]
exten = 1000,1,NoCDR()
exten = 1000,2,Answer()
exten = 1000,3,VoicemailMain(${CALLERIDNUM})
Looks at the console log again. You should be seeing
VoicemailMain(1235551212) or whatever telephone number you are calling
from. Is the telephone number you are calling from the same as the
mailbox name in voicemail.conf?
At the console I see:
Executing VoiceMailMain(SIP/pstn-1270-0819a1f0, pstn1270) in new
stack
-- Playing 'vm-login' (language 'en')
-- Playing 'vm-password' (language 'en')
-- Incorrect password '123' for user 'tn12701' (context = default)
Maybe the problem is that the user tn12701 is not in the
voicemail.conf?
I don't think it is possible to assign two users to one mail box, is
it?. I have to make two entries, I think.
What I don't understand is why the CALLERIDNUM is pstn1270. Also, you
can see the VoicemailMain is stripping off the ps, I think that may be
because you do not have a | or a , after ${CALLERIDNUM}. Why not
just REMOVE the ${CALLERIDNUM} and let VoicemailMain prompt you for the
mailbox number.
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Re: [asterisk-users] voicemail usernames can't begin with j letter?
Thanks to all that replayed, I made like Mr Watkins told me, and my problem is
apparently solved, although, because of the usage of the syntax
VoiceMail(${EXTEN}|u), now, two more sound files are played: vm-theperson and
vm-isunavail, while before were only played vm-intro and beep.
Is there a way to disable this two other files that get played every time?
see http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+VoiceMail
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Re: [asterisk-users] Audiocodes MP-20x
The reason not many people have this product, is because this product is not going to be available to the public at this time.Audiocodes will only provide this product (currently due to ship in December) as 1,000-piece minimum orders for the MP-202. The MP-201 will be available sometime quarter 1 2007 and then the mixed FXS/FXO 202 will follow. MSRP is currently estimated at $99/unit.This unit is only to be sold to service providers and large installs per Audiocodes current VoIP direction they are moving.If you'd like more information on obtaining / testing this unit, you can contact me off list. Jessee HolmesAtacomm / Ataractic Corporationwww.atacomm.comV: 1-877-700-VOIP[EMAIL PROTECTED]Looking for voice over IP products? Visit our VoIP store at http://voipstore.atacomm.com/ On Oct 22, 2006, at 5:47 PM, Andrew Joakimsen wrote:Has anyone used the AudioCodes MP-20x? http://audiocodes.com/Objects/Analog_Telephone_Adapter_Series_MP_20X.pdfSeems like a good device, but I can't seem to find anyone actually using them... ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] INVAL Messages
All, Has anyone seen INVAL messages on an IAX link before? I'm occasionally gettingthem from my Gateway provider, and I need to narrow down the potential cause. Symptoms are: Incoming calls fail, I see NEW, AUTHREQ then INVAL messages between the two A*k boxes... then for no reason at all it'll start working ok again.. My Asterisik: 1.2.10, Gateway A*k : 1.2.0 - Any known issues with IAX on either? My best guess so far is that the packets are getting corrupted on-route.. and I've asked the gateway folks to capture the traffic when it happens again to confirm... Thanks, Adrian ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] voicemail usernames can't begin with j letter?
Thanks for all that replayed, the problem is solved! Regards, Ricardo. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CID Issues
The o option is mentioned over at FreePBX and how to restore this setting. On 10/23/06, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: mail-lists wrote: Hello, I've posted this at the trixbox and freepbx forums and haven't been able to get an answer. I thought perhaps the guru's here might be able to help me out :) I'm having some issues with setting caller IDs. There are 2 problems that I would like to solve. 1. I have a DID pointing to a ring group. The only 'extension' in that ring group is an external number (cell phone). So essentially the DID fwds to a cell phone. The problem is that the CID that shows up on the Cell phone is the number that's set on the outgoing trunk the the CALLERS #. Is there a simple way to override this? Or better yet, is there a prefered method for forwarding calls out with freePBX? I cannot help you with Trixbox, only Asterisk.Are you using PRI, Analog, or VoIP for your outgoing call?If PRI or VoIP AND the carrier permits it, you can manually set theCaller*ID before the Dial() line using SetCIDNum. You can also do a show application dial in the Asterisk CLI, payspecial attention to the o option to Dial()___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users -- Tom VileBaldwin Technology Solutions, IncConsulting - Web Design - VoIP Telephonywww.baldwintechsolutions.com Phone: 518-631-2855 x205Fax: 518-631-2856 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: checking 'voicemail externally - doesn't work
[snip]
What I don't understand is why the CALLERIDNUM is pstn1270. Also, you
can see the VoicemailMain is stripping off the ps, I think that may be
because you do not have a | or a , after ${CALLERIDNUM}. Why not
just REMOVE the ${CALLERIDNUM} and let VoicemailMain prompt you for the
mailbox number.
Caller ID pstn1270 is coming from Sipura PSTN line.
Though, I'm very confused whey is it truncating the caller id from
pstn1270 to 'tn127011' (see below); where is it getting it from ???
-- Executing VoiceMailMain(SIP/pstn-1270-081a4c70, pstn1270|) in
new stack
-- Playing 'vm-login' (language 'en')
-- Playing 'vm-password' (language 'en')
-- Incorrect password '123' for user 'tn127011' (context = default
Adding | pipe to the context doesn't help, without ${CALLERIDNUM} is
just an inconvenience as I have to enter voice mail box number.
exten = 1000,3,VoicemailMain(${CALLERIDNUM}|)
Though, I've performed an experiment, added s - no password
exten = 1000,3,VoicemailMain(${CALLERIDNUM}|s)
so now calling internally works, I don't have to enter password.
Calls coming externally from pstn line, it worked but ask me for
password (see below):
vm_execmain: Specified user 'pstn1270' not found (check voicemail.conf
and/or realtime config). Falling back to authentication mode.
At this point it is asking me for password and it accept it.
But when I remove the s the password is not going through, I think
because it is changing somehow, the caller ID from pstn1270 to
'tn127011' and I don't know why?
Could it be a bug?
--
#Joseph
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[asterisk-users] call file mechanism
Hi list, I have a call file as following and it works. But, I don't really understand its mechanism. The SIP/voipbuster is a sip trunk which I set up in freePBX with voipbuster account. And 2874 is one of my extension which was assigned to x-lite client. When I place this call file in outgoing folder, it is able to dial out my home phone at 001xx. However, the Dst in call logs show 2874 or s instead of my phone number. Why sometimes 2874, sometimes s? and why not my phone number? My interpretation is the call file actually call extension 2874 and place a out going call via 2874. If I am right, does it mean any outgoing call has to be placed through an extension. How can I manipulate this call file in order to show my home phone as destination instead of extension number. Thank you very much. Channel: SIP/voipbuster/001xx MaxRetries: 1 RetryTime: 60 WaitTime: 30 Context: outgoing Extension: 2874 Priority: 1 Thanks in advance!! _ Try the next generation of search with Windows Live Search today! http://imagine-windowslive.com/minisites/searchlaunch/?locale=en-ussource=hmtagline ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] INVAL Messages
Hi Adrian, are you using this IAX thru NAT? I have this problem when i try call with IAX2 and this Asterisk server is behind the NAT... I think its here problem with UDP source port which is changed in NAT router, but im not sure 100% Marian Adrian Marsh napsal(a): All, Has anyone seen INVAL messages on an IAX link before? I'm occasionally getting them from my Gateway provider, and I need to narrow down the potential cause. Symptoms are: Incoming calls fail, I see NEW, AUTHREQ then INVAL messages between the two A*k boxes... then for no reason at all it'll start working ok again.. My Asterisik: 1.2.10, Gateway A*k : 1.2.0- Any known issues with IAX on either? My best guess so far is that the packets are getting corrupted on-route.. and I've asked the gateway folks to capture the traffic when it happens again to confirm... Thanks, Adrian ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Marian Rychtecky [EMAIL PROTECTED] Tel. +420 724 397 441 ICQ 76582857 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Audiocodes MP-20x
If you dont mind me asking a few questions, I am wondering, to what extent have you tested the units? Do all the basic functions (call id, call waiting, call transfer, forwarding, etc) work on the unit? How well do the router functions work? Overall quality and impressions? On 10/23/06, Andrew Nowrot [EMAIL PROTECTED] wrote: Hi Has anyone used the AudioCodes MP-20x?I've been testing this for 3 weeks now. No problems so far. This gateway has many features including IPSec and is not that expensive. RegardsAndrew ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] checking 'voicemail externally - doesn't work
Second authentication DISA is for additional security Actually that's called paranioa
and it doesn'tcause any problem, Obviously you do have a problem accessing your voicemail, else you would not be posting this
the authentication is giving me access to voicemailbut password is not recognized.
It's giving you access to the voicemailmain application. What is the mailbox? Is it PSTN3434 like your caller ID? If you are calling in from (presumably) any phone does it make sense to use the caller id as the mailbox number? Why not use this:
[voicemail]exten = 1000,1,NoCDR()exten = 1000,2,Answer()exten = 1000,3,VoicemailMain(${CALLERIDNUM})
[disa-access]include = tollfreeexten = 1000,1,VoicemailMain()You will have to enter the mailbox number now as well as the password, so instead of being protected by two passwords it is not protected by two passwords AND the mailbox number. So you can remove the disa and the hacker still has to know two unique numbers to access the voicemail... same amount of security.
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Re: [asterisk-users] asterisk guru needed for job in Chicago area
Elvar Are you looking at a multi-site VoIP system or just replacing a PBX at a large site. Using VoIP at a multi-site client that does not have full time service personnel can lead to a failure that affects business operations. Always verify your COOP before starting such an install. Please remember that people care more about phones than computers, the receptionist is your friend. Andrew COOP = http://en.wikipedia.org/wiki/Continuity_of_Operations_Plan On 10/23/06, Elvar [EMAIL PROTECTED] wrote: Hello, I run a small network consulting company in the Chicago area and I have a client who is interested in doing an asterisk based VOIP installation. My company does not have the necessary experience to carry out the project alone so I am looking for an asterisk guru to lead the project. I'm interested in someone from the Chicago or northwest Indiana area who is very experienced with Asterisk deployements in multi-site scenarios connected via VPN tunnels. The person must be very experienced with the following; - Working with various telcos to order and troubleshoot circuits and phone lines - Analog based VOIP gateways - Asterisk PBX on Linux - VOIP in general - SIP and IAX VOIP protocols - Solid experience with IP networks, routers, switches, firewalls The person must also be willing to come on site during deployement to ensure smooth integration but a good portion of the work may possibly be done remotely since we can handle some of it. This is for a one project job initially but if it goes well it could definitely open the door for other VOIP related projects. For anyone who might be interested, please email me your resume. Kind regards, Elvar ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- --- Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) [EMAIL PROTECTED] - [EMAIL PROTECTED] If any of the above are down we have bigger problems than my email! Hind sight is most always 20/20 or better. --- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] One way audio half way through call
Hi, I have asterisk 1.2.12 running on my server. Everything seems to be working fine on it. It has an IAX connection to the terminator/orignator. Again, everything seems to be fine.. calls come in and go out. However, it seems that after a call has been up for several minutes audio will go one-way. That is, we can hear the other person, but they can not hear us. Any thoughts? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] REQ: Astricon Pictures
Anybody with photo's (for this astricon or any asterisk related event), please upload them at: http://www.asteriskguru.com/gallery/main.php It's possible to upload as a guest without registering, if somebody sees kiddie porn etc, please warn me so that i can disable this. I will be adding some myself later today. Zoa. P.S. Free beer for everybody who makes pictures with Matt Fredrickson dangling upside down ( http://www.asteriskguru.com/gallery/main.php?g2_itemId=26 ) or drinking alcohol or redbull! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk and dialer Running on Thin Clients
Hi everybody Im the IT Manager for a new call center and my bosses has assing to me a very dificult task i have to configure the call center using Hp 5520 thin clients, asterisk and some kind of dialer that allows outbound calls. I triyed using terminal services but it dind worked because the lack on the sound and the microphone do not work on the thin clients using terminal services, we tried to install Linux in the Thin Clients but they are to small to to a have a decent OS inside, im waiting a demo version of Citrix in order to see if we canget the sofphones work i dont know what else to tink because in top of tha we nee to get a dialer tha supports this enviroment. PLEASE ANY HELP WILL BE MORE THAN WELCOME.. Ignacio. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Question on one-way-audio with IAX
Hi, I have asterisk 1.2.12 running on my server. Everything seems to be working fine on it. It has an IAX connection to the terminator/orignator. Again, everything seems to be fine.. calls come in and go out. However, it seems that after a call has been up for several minutes audio will go one-way. That is, we can hear the other person, but they can not hear us. Any thoughts? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom provision errors still! Arg!
I have been struggling over central provisioning for quite some time. I have eagerly watched each post with like problems but have yet to find a reliable answer. I have a Polycom 501 and I am trying to provision from an FTP server, and just to take any routing out of the issue it is on the same subnet. I am running the 2.0.1.0291 firmware and 3.2.2 bootrom. I set the IP info on the phone and point it at the ftp server. It successfully loaded the new firmware and bootrom but will not provision. Every time it gives me Config file error: The error is 0x0 after the page that says Processing Configuration This may take a minute. Here is my ftp log: Mon Oct 23 11:53:18 2006 1 x.x.x.x 339 /home/pcom/0004f2027255.cfg b _ o r pcom ftp 0 * c Mon Oct 23 11:53:19 2006 1 x.x.x.x 10240 /home/pcom/sip.ld b _ o r pcom f tp 0 * i Mon Oct 23 11:53:19 2006 1 x.x.x.x 0 /home/pcom/x102\x102.cfg b _ o r pco m ftp 0 * i Mon Oct 23 11:53:27 2006 6 x.x.x.x 121872 /home/pcom/sip.cfg b _ o r pcom ftp 0 * c Mon Oct 23 11:54:07 2006 1 x.x.x.x 9638 /home/pcom/x102/0004f2027255-boot .log b _ i r pcom ftp 0 * c Here is the boot log: |-- Initial log entry -- 1023201556|so |4|00|+++ Note that bootrom log times are in GMT +++ 1023201556|hw |4|00|Initial log entry. 1023201556|wdog |4|00|Initial log entry 1023201556|cfg |4|00|Initial log entry 1023201556|copy |3|00|Initial log entry 1023201556|cdp |4|00|Initial log entry 1023201556|cdp |5|00|CDP is DISABLED. 1023201556|cdp |5|00|802.1Q/VLAN tagging is DISABLED. 1023201556|so |3|00|Platform: Model=SoundPoint IP 501, Assembly=2345-11500-040 Rev=A 1023201556|so |3|00|Platform: Board=2345-11500-040 A 1023201556|so |3|00|Platform: MAC=0004f2027255, IP=172.16.27.10, Subnet Mask=255.255.255.224 1023201556|so |3|00|Platform: BootBlock=2.5.0 (11500_040) 06-Nov-04 08:08 1023201556|so |3|00|Application, main: Label=BOOT, Version=3.2.2.0019 24-Aug-06 18:05 1023201556|so |3|00|Application, main: P/N=3150-11069-322 1023201556|app1 |4|00|Initial log entry. 1023201556|app1 |3|00|DNS resolver servers are 'x.x.x.x' x.x.x.x' 1023201556|app1 |3|00|DNS resolver search domain is '' 1023201556|app1 |3|00|Bootline: eim(0,0)bootHost:flash e=172.16.27.10:ffe0 h=172.16.27.6 g=172.16.27.1 u=pcom pw= tn=CircaIP 1023201827|app1 |3|00|Time has been set from x.x.x.x (x.x.x.x). 1023201827|so |3|00|Link status is Net up Speed 100 full Duplex, PC up Speed 100 full Duplex. 1023201833|cfg |3|00|Beginning to provision phone 1023201833|copy |3|00|'ftp://pcom:[EMAIL PROTECTED]/bootrom.ld' from '172.16.27.6' 1023201903|cfg |3|00|Image bootrom.ld has not changed 1023201903|copy |3|00|Download of 'bootrom.ld' succeeded on attempt 1 (addr 1 of 1) 1023201903|cfg |3|00|Downloaded bootROM is identical to Current version 3.2.2 1023201903|copy |3|00|'ftp://pcom:[EMAIL PROTECTED]/0004f2027255.cfg' from '172.16.27.6' 1023201939|copy |3|00|Download of '0004f2027255.cfg' succeeded on attempt 1 (addr 1 of 1) 1023201939|copy |3|00|'ftp://pcom:[EMAIL PROTECTED]/sip.ld' from '172.16.27.6' 1023202009|cfg |3|00|Image sip.ld has not changed 1023202009|copy |3|00|Download of 'sip.ld' succeeded on attempt 1 (addr 1 of 1) 1023202009|cfg |3|00|Downloaded application image is identical to current version 1023202009|cfg |3|00|Phone successfully provisioned 1023202041|app1 |4|00|Loaded application sip.ld successfully, errors 0x0. 1023202041|app1 |6|00|Uploading boot log, time is MON OCT 23 20:20:42 2006 And it repeats this every time. I can provide the sip.cfg and mac.cfg on request. I dont want to run out of space for the post. Please help! It really cant be this hard. Curt ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] checking 'voicemail externally - doesn't work
On Mon, 2006-10-23 at 15:34 -0400, Andrew Joakimsen wrote:
Second authentication DISA is for additional security
Actually that's called paranioa
I just try with single authentication DISA, doesn't work, password is
not recognized.
[snip]
the authentication is giving me access to voicemail
but password is not recognized.
It's giving you access to the voicemailmain application. What is the
mailbox? Is it
Mailbox number is 11
PSTN3434 like your caller ID? If you are calling in from (presumably)
any phone does it make sense to use the caller id as the mailbox
number? Why not use this:
[voicemail]
exten = 1000,1,NoCDR()
exten = 1000,2,Answer()
exten = 1000,3,VoicemailMain(${CALLERIDNUM})
[disa-access]
include = tollfree
exten = 1000,1,VoicemailMain()
It only works if I disable password with |s
exten = 1000,3,VoicemailMain(${CALLERIDNUM}|s)
In this case all internal callers can access their voicemailbox without
password but when a call comes from an external source PSTN line it is
asking for password and it goes through correctly:
vm_execmain: Specified user 'pstn1270' not found (check voicemail.conf
and/or realtime config). Falling back to authentication mode.
(as the user pstn1270 is not in voicemail.conf file)
but without the |s somehow it is distorting the caller ID from
pstn1270 to
'tn127011' that is why it doesn't work, but I can not pin-point what is
changing caller ID.
--
#Joseph
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[asterisk-users] CBeyond SIP
It looks like the deal CBeyond is offering me for a T1 to the office and VoIP service via SIP is going to win in my current effort to get away from the local telco. The idea of using a VoIP carrier with QoS all the way between here and there and back is very appealing after working on an ADSL line for over a year. My question is whether or not I'm going to have any troubles connecting to their central Cisco call director (I think) SIP servers? I'd expect not but figured I'd ask if anyone else was using them just in case. Paul Paul Dugas Computer Engineer Dugas Enterprises, LLC 522 Black Canyon Park Canton, GA 30114 phone: 404.932.1355 fax: 866.751.6494 [EMAIL PROTECTED] http://DugasEnterprises.com This e-mail and any attachments are confidential. If you receive this message in error or are not the intended recipient, you should not retain, distribute, disclose or use any of this information and you should destroy the e-mail and any attachments or copies. signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Where to best start looking for voicemail/moh sound quality problem?
I'm running Asterisk 1.2.13 on a Solaris 10 X86 box behind an IPCop firewall on a 5Mbps down/512 up cable connection. I'm having sound quality problems when users call in for voicemail and with music on hold. The sound is choppy and muffled while souding pretty good for calls inside the network. I'd appreciate some pointers as to where to start looking to improve things. I've tried setting QOS paramters for IPCop but I'm sure that had any effect. Frank ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Where to best start looking for voicemail/moh sound quality problem?
Do you have the issues locally ? Are you using Ztdummy ? - Original Message - From: Frank Tarczynski [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Monday, October 23, 2006 10:48 PM Subject: [asterisk-users] Where to best start looking for voicemail/moh sound quality problem? I'm running Asterisk 1.2.13 on a Solaris 10 X86 box behind an IPCop firewall on a 5Mbps down/512 up cable connection. I'm having sound quality problems when users call in for voicemail and with music on hold. The sound is choppy and muffled while souding pretty good for calls inside the network. I'd appreciate some pointers as to where to start looking to improve things. I've tried setting QOS paramters for IPCop but I'm sure that had any effect. Frank ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Multiple line phones with different contexts
Hey all,
Has anyone had any issues with phones having multiple lines that are in
different contexts? We've got a couple phones that we're testing
intercom functionality for, and I'm noticing that for some strange
reason, no matter what line we use, the phones tend to be completely in
one context or another, not segregated like I would expect.
Our contexts look like this:
context intercom {
_ = {
Answer;
check-cid();
Set(CALLERID(num)=${CALLERID(num)} (INT));
SIPAddHeader(Alert-Info: Ring Answer);
createds(${EXTEN});
Dial(SIP/${ds}|20);
Hangup;
};
};
context long-distance {
includes {
local;
};
_9011 = dialout(${EXTEN});
_91NXXNXX = dialout(${EXTEN});
};
The phones are configured as such:
[0004F2100526_1]
canreinvite=no
context=long-distance
host=dynamic
nat=no
qualify=6
secret=secret
type=peer
regexten=44198
[0004F2100526_2]
canreinvite=no
context=intercom
host=dynamic
nat=no
qualify=6
secret=secret
type=peer
regexten=44198
A sip debug from one of the intercoms:
-- SIP read from 10.20.136.130:5060:
INVITE sip:[EMAIL PROTECTED]:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.20.136.130;branch=z9hG4bK2a5fd1d91B78BACE
From: Aaron Daniel
sip:[EMAIL PROTECTED];tag=DDF0722-FFF8D457
To: sip:[EMAIL PROTECTED];user=phone
CSeq: 1 INVITE
Call-ID: [EMAIL PROTECTED]
Contact: sip:[EMAIL PROTECTED]
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE,
NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_430-UA/2.0.1.0291
Supported: 100rel,replaces
Allow-Events: talk,hold,conference
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 251
v=0
o=- 1161637564 1161637564 IN IP4 10.20.136.130
s=Polycom IP Phone
c=IN IP4 10.20.136.130
t=0 0
a=sendrecv
m=audio 2240 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
--- (14 headers 11 lines)---
Using INVITE request as basis request -
[EMAIL PROTECTED]
Sending to 10.20.136.130 : 5060 (non-NAT)
Found peer '0004F2100526_1'
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 10.20.136.130:2240
Found description format PCMU
Found description format PCMA
Found description format G729
Found description format telephone-event
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x10c
(ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
Looking for 4000 in long-distance (domain tcm1.shsu.edu)
Reliably Transmitting (no NAT) to 10.20.136.130:5060:
SIP/2.0 484 Address Incomplete
Via: SIP/2.0/UDP
10.20.136.130;branch=z9hG4bK2a5fd1d91B78BACE;received=10.20.136.130
From: Aaron Daniel
sip:[EMAIL PROTECTED];tag=DDF0722-FFF8D457
To: sip:[EMAIL PROTECTED];user=phone;tag=as04c17ab8
Call-ID: [EMAIL PROTECTED]
CSeq: 1 INVITE
User-Agent: SCM1 - Sip Call Manager 1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: sip:[EMAIL PROTECTED]
Content-Length: 0
---
tcm1*CLI
-- SIP read from 10.20.136.130:5060:
ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.20.136.130;branch=z9hG4bK2a5fd1d91B78BACE
From: Aaron Daniel
sip:[EMAIL PROTECTED];tag=DDF0722-FFF8D457
To: sip:[EMAIL PROTECTED];user=phone;tag=as04c17ab8
CSeq: 1 ACK
Call-ID: [EMAIL PROTECTED]
Contact: sip:[EMAIL PROTECTED]
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE,
NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_430-UA/2.0.1.0291
Max-Forwards: 70
Content-Length: 0
--- (11 headers 0 lines)---
Destroying call '[EMAIL PROTECTED]'
Finally, a sip show peer on the intercom line proving asterisk knows
it's in the right context:
tcm1*CLI sip show peer 0004F2100526_2
tcm1*CLI
* Name : 0004F2100526_2
Secret : Set
MD5Secret: Not set
Context : intercom
Subscr.Cont. : Not set
Language :
AMA flags: Unknown
CallingPres : Presentation Allowed, Not Screened
Callgroup:
Pickupgroup :
Mailbox : [EMAIL PROTECTED]
VM Extension : asterisk
LastMsgsSent : 0
Call limit : 0
Dynamic : Yes
Callerid :
Expire : 2252
Insecure : port,invite
Nat : RFC3581
ACL : No
CanReinvite : No
PromiscRedir : No
User=Phone : No
Trust RPID : No
Send RPID: Yes
DTMFmode : rfc2833
LastMsg : 0
ToHost :
Addr-IP : 10.20.136.130 Port 5060
Defaddr-IP : 0.0.0.0 Port 5060
Def. Username: 0004F2100526_2
SIP Options : (none)
Codecs : 0x8000e (gsm|ulaw|alaw|h263)
Codec Order : (none)
Status : OK (14 ms)
Useragent: PolycomSoundPointIP-SPIP_430-UA/2.0.1.0291
Reg. Contact : sip:[EMAIL PROTECTED]
ANY help would be greatly
Re: [asterisk-users] Asterisk and dialer Running on Thin Clients
Sell the thin clients and their home server on Ebay and buy some used desktops and an Asterisk server. VOIP with softphones on thin clients does not work very well at all unless you seriously limit the number of clients attached to each server. How many seats is this supposed to be? MATT--- On 10/23/06, Ignacio Ortega A. [EMAIL PROTECTED] wrote: Hi everybody Im the IT Manager for a new call center and my bosses has assing to me a very dificult task i have to configure the call center using Hp 5520 thin clients, asterisk and some kind of dialer that allows outbound calls. I triyed using terminal services but it dind worked because the lack on the sound and the microphone do not work on the thin clients using terminal services, we tried to install Linux in the Thin Clients but they are to small to to a have a decent OS inside, im waiting a demo version of Citrix in order to see if we can get the sofphones work i dont know what else to tink because in top of tha we nee to get a dialer tha supports this enviroment. PLEASE ANY HELP WILL BE MORE THAN WELCOME.. Ignacio. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and dialer Running on Thin Clients
100 for start and 400 in february OH GOD !! On 10/23/06, Matt Florell [EMAIL PROTECTED] wrote: Sell the thin clients and their home server on Ebay and buy some useddesktops and an Asterisk server. VOIP with softphones on thin clients does not work very well at allunless you seriously limit the number of clients attached to eachserver.How many seats is this supposed to be?MATT---On 10/23/06, Ignacio Ortega A. [EMAIL PROTECTED] wrote: Hi everybody Im the IT Manager for a new call center and my bosses has assing to me a very dificult task i have to configure the call center using Hp 5520 thin clients, asterisk and some kind of dialer that allows outbound calls. I triyed using terminal services but it dind worked because the lack on the sound and the microphone do not work on the thin clients using terminal services, we tried to install Linux in the Thin Clients but they are to small to to a have a decent OS inside, im waiting a demo version of Citrix in order to see if we can get the sofphones work i dont know what else to tink because in top of tha we nee to get a dialer tha supports this enviroment. PLEASE ANY HELP WILL BE MORE THAN WELCOME.. Ignacio. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and dialer Running on Thin Clients
SO THERES NOT WAY TO MAKE VOIP WITH THIN CLIENTS YOU SAID? On 10/23/06, Ignacio Ortega A. [EMAIL PROTECTED] wrote: 100 for start and 400 in february OH GOD !! On 10/23/06, Matt Florell [EMAIL PROTECTED] wrote: Sell the thin clients and their home server on Ebay and buy some useddesktops and an Asterisk server. VOIP with softphones on thin clients does not work very well at allunless you seriously limit the number of clients attached to eachserver.How many seats is this supposed to be?MATT---On 10/23/06, Ignacio Ortega A. [EMAIL PROTECTED] wrote: Hi everybody Im the IT Manager for a new call center and my bosses has assing to me a very dificult task i have to configure the call center using Hp 5520 thin clients, asterisk and some kind of dialer that allows outbound calls. I triyed using terminal services but it dind worked because the lack on the sound and the microphone do not work on the thin clients using terminal services, we tried to install Linux in the Thin Clients but they are to small to to a have a decent OS inside, im waiting a demo version of Citrix in order to see if we can get the sofphones work i dont know what else to tink because in top of tha we nee to get a dialer tha supports this enviroment. PLEASE ANY HELP WILL BE MORE THAN WELCOME.. Ignacio. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Strange Zaptel Problem
I am running a call center with 20-30 operators with outbound projects. We have an Digium Quad port E1 interface (TE410P) on an IBM Server running Ubuntu-server with the lastest version of Asterisk Zaptel and Libpri. The problem is that when there are about 15 or more active calls on the zap interface asterisk start behaving in a weird way: it starts to lock for a second or two every few seconds (if i enter reload it just display the prompt again and the reload executes 2 or 3 seconds later) but the most pressing problem is that the sip softphones that the operators use cannot register randomly (but very often) and the call initialization takes a very long time (30 -45 sec) and sometimes times out. The more channels are in use the worst it gets. If there are no zap channels in use everything works perfectly in the SIP side. I am suspecting that during the time asterisk is locked up it looses sip udp packets. but the weird thing is that established calls are not affected (RTP works fine). If you have any ideas please reply. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] checking 'voicemail externally - doesn't work
On 10/23/06, Joseph [EMAIL PROTECTED] wrote:
I just try with single authentication DISA, doesn't work, password isnot recognized.Try without any disa whatsoever
[snip] the authentication is giving me access to voicemail but password is not recognized. It's giving you access to the voicemailmain application. What is the mailbox? Is it
Mailbox number is 11PSTN3434 like your caller ID? If you are calling in from (presumably) any phone does it make sense to use the caller id as the mailbox number? Why not use this:
[voicemail] exten = 1000,1,NoCDR() exten = 1000,2,Answer() exten = 1000,3,VoicemailMain(${CALLERIDNUM}) [disa-access]
include = tollfree exten = 1000,1,VoicemailMain()It only works if I disable password with |sexten = 1000,3,VoicemailMain(${CALLERIDNUM}|s)
Did you try exten = 1000,1,VoicemailMain() as I said above with NOTHING BETWEEN THE PARENTHASIS???
In this case all internal callers can access their voicemailbox withoutpassword but when a call comes from an external source PSTN line it isasking for password and it goes through correctly:vm_execmain: Specified user 'pstn1270' not found (check
voicemail.confand/or realtime config).Falling back to authentication mode.(as the user pstn1270 is not in voicemail.conf file)but without the |s somehow it is distorting the caller ID from
pstn1270 to'tn127011' that is why it doesn't work, but I can not pin-point what ischanging caller ID.You said the mailbox number is 11 and the caller ID Is correctly pstn1270 and incorrectly tn127011 since the mailbox number is 11, I don't see how fixing (what does your CDR say??) this issue will fix your voicemail issue. Why do you insist on using the caller ID? Remember what you are trying to do, if user has to dial into the system from an outside phone their CALLER ID WILL NOT BE THEIR MAILBOX NUMBER.
For the last time, try:exten = 1000,1,VoicemailMain()inside your disa-access context, and get rid of the old voicemail include statement. That will work, here is a detailed sequence of events
Enter disa password, press #At the dial tone dial 1000System says Comedian Mail. Mailbox?You dail the mailbox number which you stated above is 11 So press the 1 key on your telephone, if you wish you can dial # after, if not just wait.
System says Password?You dial the password, if you want you can press # after it, if not just waitI'm not going to respond to this thread any more. I've given you step by step EXACTLY what to do, anyone else would have gotten a USD 100 ++ bill for that advice.
___
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Re: [asterisk-users] Where to best start looking for voicemail/moh sound quality problem?
Do you use a VoIP provider? Is that provider by any chance VoicePulse? Have you tried other providers?You can get free DID's at http://www.ipkall.com/ and http://www.trxtel.com/ that would be a good place to start if you don't have any more providers...On 10/23/06, Frank Tarczynski [EMAIL PROTECTED] wrote:I'm running Asterisk 1.2.13 on a Solaris 10 X86 box behind an IPCop firewall on a 5Mbps down/512 up cable connection.I'm having sound quality problems when users call in for voicemail andwith music on hold.The sound is choppy and muffled while souding prettygood for calls inside the network. I'd appreciate some pointers as to where to start looking to improve things.I've tried setting QOS paramters for IPCop but I'm sure that had any effect.Frank___ --Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom provision errors still! Arg!
What if you just use the default configuration files?On 10/23/06, Curt Shaffer [EMAIL PROTECTED] wrote: I have been struggling over central provisioning for quite some time. I have eagerly watched each post with like problems but have yet to find a reliable answer. I have a Polycom 501 and I am trying to provision from an FTP server, and just to take any routing out of the issue it is on the same subnet. I am running the 2.0.1.0291 firmware and 3.2.2 bootrom. I set the IP info on the phone and point it at the ftp server. It successfully loaded the new firmware and bootrom but will not provision. Every time it gives me Config file error: The error is 0x0 after the page that says Processing Configuration This may take a minute. Here is my ftp log: Mon Oct 23 11:53:18 2006 1 x.x.x.x 339 /home/pcom/0004f2027255.cfg b _ o r pcom ftp 0 * c Mon Oct 23 11:53:19 2006 1 x.x.x.x 10240 /home/pcom/sip.ld b _ o r pcom f tp 0 * i Mon Oct 23 11:53:19 2006 1 x.x.x.x 0 /home/pcom/x102\x102.cfg b _ o r pco m ftp 0 * i Mon Oct 23 11:53:27 2006 6 x.x.x.x 121872 /home/pcom/sip.cfg b _ o r pcom ftp 0 * c Mon Oct 23 11:54:07 2006 1 x.x.x.x 9638 /home/pcom/x102/0004f2027255-boot .log b _ i r pcom ftp 0 * c Here is the boot log: |-- Initial log entry -- 1023201556|so |4|00|+++ Note that bootrom log times are in GMT +++ 1023201556|hw |4|00|Initial log entry. 1023201556|wdog |4|00|Initial log entry 1023201556|cfg |4|00|Initial log entry 1023201556|copy |3|00|Initial log entry 1023201556|cdp |4|00|Initial log entry 1023201556|cdp |5|00|CDP is DISABLED. 1023201556|cdp |5|00|802.1Q/VLAN tagging is DISABLED. 1023201556|so |3|00|Platform: Model=SoundPoint IP 501, Assembly=2345-11500-040 Rev=A 1023201556|so |3|00|Platform: Board=2345-11500-040 A 1023201556|so |3|00|Platform: MAC=0004f2027255, IP=172.16.27.10, Subnet Mask= 255.255.255.224 1023201556|so |3|00|Platform: BootBlock=2.5.0 (11500_040) 06-Nov-04 08:08 1023201556|so |3|00|Application, main: Label=BOOT, Version=3.2.2.0019 24-Aug-06 18:05 1023201556|so |3|00|Application, main: P/N=3150-11069-322 1023201556|app1 |4|00|Initial log entry. 1023201556|app1 |3|00|DNS resolver servers are 'x.x.x.x' x.x.x.x' 1023201556|app1 |3|00|DNS resolver search domain is '' 1023201556|app1 |3|00|Bootline: eim(0,0)bootHost:flash e=172.16.27.10:ffe0 h=172.16.27.6 g= 172.16.27.1 u=pcom pw= tn=CircaIP 1023201827|app1 |3|00|Time has been set from x.x.x.x (x.x.x.x). 1023201827|so |3|00|Link status is Net up Speed 100 full Duplex, PC up Speed 100 full Duplex. 1023201833|cfg |3|00|Beginning to provision phone 1023201833|copy |3|00|' ftp://pcom:[EMAIL PROTECTED]/bootrom.ld' from '172.16.27.6' 1023201903|cfg |3|00|Image bootrom.ld has not changed 1023201903|copy |3|00|Download of 'bootrom.ld' succeeded on attempt 1 (addr 1 of 1) 1023201903|cfg |3|00|Downloaded bootROM is identical to Current version 3.2.2 1023201903|copy |3|00|'ftp://pcom:[EMAIL PROTECTED]/0004f2027255.cfg' from ' 172.16.27.6' 1023201939|copy |3|00|Download of '0004f2027255.cfg' succeeded on attempt 1 (addr 1 of 1) 1023201939|copy |3|00|' ftp://pcom:[EMAIL PROTECTED]/sip.ld' from '172.16.27.6' 1023202009|cfg |3|00|Image sip.ld has not changed 1023202009|copy |3|00|Download of 'sip.ld' succeeded on attempt 1 (addr 1 of 1) 1023202009|cfg |3|00|Downloaded application image is identical to current version 1023202009|cfg |3|00|Phone successfully provisioned 1023202041|app1 |4|00|Loaded application sip.ld successfully, errors 0x0. 1023202041|app1 |6|00|Uploading boot log, time is MON OCT 23 20:20:42 2006 And it repeats this every time. I can provide the sip.cfg and mac.cfg on request. I don't want to run out of space for the post. Please help! It really can't be this hard. Curt ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and dialer Running on Thin Clients
Hi On Mon, Oct 23, 2006 at 04:08:15PM -0400, Ignacio Ortega A. wrote: Hi everybody Im the IT Manager for a new call center and my bosses has assing to me a very dificult task i have to configure the call center using Hp 5520 thin clients, asterisk and some kind of dialer that allows outbound calls. I triyed using terminal services but it dind worked because the lack on the sound and the microphone do not work on the thin clients using terminal services, we tried to install Linux in the Thin Clients but they are to small to to a have a decent OS inside, im waiting a demo version of Citrix in order to see if we can get the sofphones work Too small? They can't bee too small to run a minimal Linux with a simple X desktop and a SIP client. Note that you better not use a terminal server settings. The SIP client should run on the thin client's CPU, not on the server's CPU. The server can help with the boot process (maybe a shared NFS root will prove useful). i dont know what else to tink because in top of tha we nee to get a dialer tha supports this enviroment. PLEASE ANY HELP WILL BE MORE THAN WELCOME.. Please don't SHOUT. -- Tzafrir Cohen iax:[EMAIL PROTECTED]/tzafrir icq#16849755 mailto:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Where to best start looking for voicemail/moh sound quality problem?
Date: Mon, 23 Oct 2006 16:48:09 -0400 (EDT)From: Frank Tarczynski [EMAIL PROTECTED]Subject: [asterisk-users] Where to best start looking forvoicemail/moh sound quality problem?To: asterisk-users@lists.digium.comMessage-ID:[EMAIL PROTECTED]Content-Type: text/plain;charset=iso-8859-1 I'm running Asterisk 1.2.13 on a Solaris 10 X86 box behind an IPCopfirewall on a 5Mbps down/512 up cable connection.I'm having sound quality problems when users call in for voicemail andwith music on hold.The sound is choppy and muffled while souding pretty good for calls inside the network.I'd appreciate some pointers as to where to start looking to improve things.I've tried setting QOS paramters for IPCop but I'm sure that had any effect.Frank Frank-I'm afraid a little more detail may be necessary to help you. From this, I have to assume the users are calling in over some sort of network connection- say SIP or IAX from a provider, and not on the analog, and that they're making a single stop- at the server hosting Asterisk for MOH and Voicemail. If that's not accurate, please let me know. Given that picture, keep in mind that asymetric bandwidth is always a bad match for VoIP, and has to be managed very, very carefully. The simplest way to think of it is to use a rate limiter to make that 512/512, which will improve quality, and should still leave you plenty of room for calls, assuming a reasonable codec compression level. (not to mention, if your QOS is set up that way, you get good amount of headroom for downloads and other traffic) That being said, both MOH and Voicemail are sort of special- meaning that they're both using different audio sources, with varying levels of compression on their own. If you have high compression (gsm or 729) on your trunk, and are trying to play back high quality audio (128K mp3 MOH sources, for instance), the transcoding is bound to produce some 'garble'. This is the reason hold music is a bad choice for playback via a cellphone- it almost always sounds like garbage. Your voicemail problems- is the quality of the prompts bad, or the recorded message itself? Both, and either, have different audio characteristics, and compression applies differently. So, to start, try switching your MOH source to an encoding that matches the trunk- depending on your version of Asterisk and how you've got MOH configured, this can be done a lot of different ways- check the wiki for it. See also if you can match the recording of voicemail to your codec, or at least find one with a very low translation cost- this may help as well. For example, if your trunk is using the gsm codec, make sure you're recording voicemails as gsm, not WAV, wav49, or wav- those can be used as well, but gsm should be in the mix for best quality. -pbd ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] rxfax problem
You are using bad software to view the faxes. In Windows the picture and fax viewer seems to work fine, however in Linux KGhostView or whever the default program is does not work, however you should try KFaxView.Steve: I'm wondering if one day span_dsp will support T38, say we have a SIP provider that supports T.38 we should be able to recieve a good fax? Right now the fax is distorted a bit, I think because it does not support ECM?On 10/20/06, Mohammad Shokuie [EMAIL PROTECTED] wrote:Hi Steve, As a matter of fact, you've done a greate job in writting this library, nodoubts. I really dont know rxgain = 12 makes that much distortion but I'mcurios to know if I pass through the incoming fax to an analog fax machine on another fxs line, the machine wouldn't receive the fax too?Anyways, let me take the most benefit as im sure you'd read this post, ihave problem with the size of received page which is shrinked, can u give me a hint about this problem too :)Thanks.---M. Shokuie NiaFrom: Steve Underwood [EMAIL PROTECTED]Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.comTo: Asterisk Users Mailing List - Non-CommercialDiscussion asterisk-users@lists.digium.comSubject: Re: [Asterisk-Users] rxfax problemDate: Fri, 20 Oct 2006 20:20:18 +0800MIME-Version: 1.0Received: from lists.digium.com ([69.16.138.164]) bybay0-mc6-f10.bay0.hotmail.com with Microsoft SMTPSVC(6.0.3790.2444); Fri,20 Oct 2006 05:42:01 -0700 Received: from digium-69-16-138-164.phx1.puregig.net (localhost[127.0.0.1])by lists.digium.com (Postfix) with ESMTP id EFF0B2FC87C;Fri, 20Oct 2006 05:20:37 -0700 (MST)Received: from psmtp.com (exprod8mx13.postini.com [ 64.18.3.113])bylists.digium.com (Postfix) with SMTP id B67A62FC82Fforasterisk-users@lists.digium.com;Fri, 20 Oct 2006 05:20:05 -0700 (MST) Received: from source ([202.14.67.92]) byexprod8mx13.postini.com([64.18.7.10]) with SMTP; Fri, 20 Oct 2006 05:20:20PDTReceived: from [ 192.168.2.50](229.166.17.210.dyn.pacific.net.hk[210.17.166.229]) by cwb.pacific.net.hkwith ESMTPid k9KCKIfs013165 for asterisk-users@lists.digium.com;Fri, 20Oct 2006 20:20:19 +0800X-Message-Info: txF49lGdW43chsCTszkrRosGSMI+inUm7kbzJdpspc0=X-Original-To: asterisk-users@lists.digium.comDelivered-To: asterisk-users@lists.digium.comUser-Agent: Mozilla Thunderbird 1.0.8-1.1.fc4 (X11/20060501)X-Accept-Language: en-us, en References: [EMAIL PROTECTED]X-pstn-levels: (S:99.9/99.9 FC:95.5390 LC:95.5390 R:95.9108P: 95.9108M:97.0282 C:98.6951 )X-pstn-settings: 3 (1.:1.) s fc lc gt3 gt2 gt1 r p m cX-pstn-addresses: from [EMAIL PROTECTED] [db-null] X-BeenThere: asterisk-users@lists.digium.comX-Mailman-Version: 2.1.5Precedence: listList-Id: Asterisk Users Mailing List - Non-CommercialDiscussion asterisk-users.lists.digium.comList-Unsubscribe:http://lists.digium.com/mailman/listinfo/asterisk-users ,mailto:[EMAIL PROTECTED]?subject=unsubscribeList-Archive: http://lists.digium.com/pipermail/asterisk-usersList-Post: mailto:asterisk-users@lists.digium.comList-Help: mailto: [EMAIL PROTECTED]?subject=helpList-Subscribe:http://lists.digium.com/mailman/listinfo/asterisk-users,mailto: [EMAIL PROTECTED]?subject=subscribeErrors-To: [EMAIL PROTECTED] Return-Path: [EMAIL PROTECTED]X-OriginalArrivalTime: 20 Oct 2006 12:42:02.0256 (UTC)FILETIME=[241ED900:01C6F445] M. Shokuie Nia wrote:Dear folk,My problem solved after two day research and try and error method ;). Itwasrelated to rxgain of the board im using. I've set the rxgain to 12 and it seems made some problem. As far as I got the spandsp is so sensitive aboutnoise on the line and because of that it couldn't hand shake with othersidewell. rxfax isn't sensitive to noise at all. At a gain of 12 you've causedoverloading and distortion, and the signal cannot be decoded. Many peopleseem to be nearly deaf. They run systems at massive gain with awful distortion, and seem content until they find something like a modem or DTMFdetection doesn't work too well.Steve___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _FREE pop-up blocking with the new MSN Toolbar - get it now!http://toolbar.msn.click-url.com/go/onm00200415ave/direct/01/ ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom provision errors still! Arg!
Do you mean .cfg and sip.cfg? Could you clarify for me please and I will try that. Thanks for the suggestion. Curt On 10/23/06, Andrew Joakimsen [EMAIL PROTECTED] wrote: What if you just use the default configuration files? On 10/23/06, Curt Shaffer [EMAIL PROTECTED] wrote: I have been struggling over central provisioning for quite some time. I have eagerly watched each post with like problems but have yet to find a reliable answer. I have a Polycom 501 and I am trying to provision from an FTP server, and just to take any routing out of the issue it is on the same subnet. I am running the 2.0.1.0291 firmware and 3.2.2 bootrom. I set the IP info on the phone and point it at the ftp server. It successfully loaded the new firmware and bootrom but will not provision. Every time it gives me Config file error: The error is 0x0 after the page that says Processing Configuration This may take a minute. Here is my ftp log: Mon Oct 23 11:53:18 2006 1 x.x.x.x 339 /home/pcom/0004f2027255.cfg b _ o r pcom ftp 0 * c Mon Oct 23 11:53:19 2006 1 x.x.x.x 10240 /home/pcom/sip.ld b _ o r pcom f tp 0 * i Mon Oct 23 11:53:19 2006 1 x.x.x.x 0 /home/pcom/x102\x102.cfg b _ o r pco m ftp 0 * i Mon Oct 23 11:53:27 2006 6 x.x.x.x 121872 /home/pcom/sip.cfg b _ o r pcom ftp 0 * c Mon Oct 23 11:54:07 2006 1 x.x.x.x 9638 /home/pcom/x102/0004f2027255-boot .log b _ i r pcom ftp 0 * c Here is the boot log: |-- Initial log entry -- 1023201556|so |4|00|+++ Note that bootrom log times are in GMT +++ 1023201556|hw |4|00|Initial log entry. 1023201556|wdog |4|00|Initial log entry 1023201556|cfg |4|00|Initial log entry 1023201556|copy |3|00|Initial log entry 1023201556|cdp |4|00|Initial log entry 1023201556|cdp |5|00|CDP is DISABLED. 1023201556|cdp |5|00|802.1Q/VLAN tagging is DISABLED. 1023201556|so |3|00|Platform: Model=SoundPoint IP 501, Assembly=2345-11500-040 Rev=A 1023201556|so |3|00|Platform: Board=2345-11500-040 A 1023201556|so |3|00|Platform: MAC=0004f2027255, IP= 172.16.27.10, Subnet Mask= 255.255.255.224 1023201556|so |3|00|Platform: BootBlock=2.5.0 (11500_040) 06-Nov-04 08:08 1023201556|so |3|00|Application, main: Label=BOOT, Version= 3.2.2.0019 24-Aug-06 18:05 1023201556|so |3|00|Application, main: P/N=3150-11069-322 1023201556|app1 |4|00|Initial log entry. 1023201556|app1 |3|00|DNS resolver servers are 'x.x.x.x' x.x.x.x' 1023201556|app1 |3|00|DNS resolver search domain is '' 1023201556|app1 |3|00|Bootline: eim(0,0)bootHost:flash e=172.16.27.10:ffe0 h= 172.16.27.6 g= 172.16.27.1 u=pcom pw= tn=CircaIP 1023201827|app1 |3|00|Time has been set from x.x.x.x (x.x.x.x). 1023201827|so |3|00|Link status is Net up Speed 100 full Duplex, PC up Speed 100 full Duplex. 1023201833|cfg |3|00|Beginning to provision phone 1023201833|copy |3|00|' ftp://pcom:[EMAIL PROTECTED]/bootrom.ld' from '172.16.27.6' 1023201903|cfg |3|00|Image bootrom.ld has not changed 1023201903|copy |3|00|Download of 'bootrom.ld' succeeded on attempt 1 (addr 1 of 1) 1023201903|cfg |3|00|Downloaded bootROM is identical to Current version 3.2.2 1023201903|copy |3|00|' ftp://pcom:[EMAIL PROTECTED]/0004f2027255.cfg' from ' 172.16.27.6' 1023201939|copy |3|00|Download of '0004f2027255.cfg' succeeded on attempt 1 (addr 1 of 1) 1023201939|copy |3|00|' ftp://pcom:[EMAIL PROTECTED]/sip.ld' from '172.16.27.6' 1023202009|cfg |3|00|Image sip.ld has not changed 1023202009|copy |3|00|Download of 'sip.ld' succeeded on attempt 1 (addr 1 of 1) 1023202009|cfg |3|00|Downloaded application image is identical to current version 1023202009|cfg |3|00|Phone successfully provisioned 1023202041|app1 |4|00|Loaded application sip.ld successfully, errors 0x0. 1023202041|app1 |6|00|Uploading boot log, time is MON OCT 23 20:20:42 2006 And it repeats this every time. I can provide the sip.cfg and mac.cfg on request. I don't want to run out of space for the post. Please help! It really can't be this hard. Curt___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Curt Shaffer,MCSA,MCSESecurity+, Network+Certified IP Telephony Sepcialist202-470-6892 (home) 202-470-6893 (Business)309-412-4809 (efax) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom SP4000 ftp problem
i recently bought an SP4000 conference phone but having problem provisioning it using ftp, every time it just hangs at Updating initial configuration... screen. when i switch it to tftp, it'll work fine. i though it was bootrom/firmware issue so i upgrade it to bootrom 3.2.2/sip 2.0.1 but it makes no difference. any thoughts? p.s. i'm using debian sarge proftpd 1.2.10 and the setting works fine w/ SP501 with bootrom 3.1.2/sip 1.6.3 -- Edwin Lam [EMAIL PROTECTED] Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [SOLVED] checking 'voicemail externally - doesn't work
On Mon, 2006-10-23 at 17:59 -0400, Andrew Joakimsen wrote: On 10/23/06, Joseph [EMAIL PROTECTED] wrote: I just try with single authentication DISA, doesn't work, password is not recognized. Try without any disa whatsoever I think DISA has to be there as it gives access to internal dial tone, isn't it? I can be without password, [snip] Did you try exten = 1000,1,VoicemailMain() as I said above with NOTHING BETWEEN THE PARENTHASIS??? Thank you, Yes It Works! It works without parenthesis. I was trying to make make it to work with one voicemail context but in this case I will create another voicemail_outside context without anything between parenthesis for outside access. exten = 1000,1,VoicemailMain() In this case all internal callers can access their voicemailbox without password but when a call comes from an external source PSTN line it is asking for password and it goes through correctly: vm_execmain: Specified user 'pstn1270' not found (check voicemail.conf and/or realtime config). Falling back to authentication mode. (as the user pstn1270 is not in voicemail.conf file) but without the |s somehow it is distorting the caller ID from pstn1270 to 'tn127011' that is why it doesn't work, but I can not pin-point what is changing caller ID. You said the mailbox number is 11 and the caller ID Is correctly pstn1270 and incorrectly tn127011 since the mailbox number is 11, I don't see how fixing (what does your CDR say??) this issue will fix your voicemail issue. Why do you insist on using the caller ID? Remember what you are trying to do, if user has to dial into the system from an outside phone their CALLER ID WILL NOT BE THEIR MAILBOX NUMBER. As I've mentioned above I was trying to get by with one [voicemail] context but I guess I'll have two. For the last time, try: exten = 1000,1,VoicemailMain() inside your disa-access context, and get rid of the old voicemail include statement. That will work, here is a detailed sequence of events Enter disa password, press # At the dial tone dial 1000 System says Comedian Mail. Mailbox? You dail the mailbox number which you stated above is 11 So press the 1 key on your telephone, if you wish you can dial # after, if not just wait. System says Password? You dial the password, if you want you can press # after it, if not just wait I'm not going to respond to this thread any more. I've given you step by step EXACTLY what to do, anyone else would have gotten a USD 100 ++ bill for that advice. Thanks Andrew for your patience. -- #Joseph ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and dialer Running on Thin Clients
Tzafrir Cohen wrote: *snipped Note that you better not use a terminal server settings. The SIP client should run on the thin client's CPU, not on the server's CPU. The server can help with the boot process (maybe a shared NFS root will prove useful). *snipped that particular unit is also supposed to be able to do PXE boot. (just fyi) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
