Re: [Asterisk-Users] Newbie question - sip.conf incoming contexts
Hi,
I'm not an expert, but as far as i know, your incoming calls will
arrive with DID in ${EXTEN}
so the only thing you need is:
exten = 1234,1,GoTo(context1,1234,1) ; example for context extension
and priority
exten = 2345,1,GoTo(context2,2345,1)
exten = 3456,1,GoTo(context3,3456,1)
Be sure that you have created context1 context2 and context3 in your
extensions.conf
And in this context1 context2 and context3 you must have handler for
1234; 2345; and 3456;
example:
[context1]
exten = 1234,1,Answer()
exten = 1234,2,Playback(vm-goodbye)
exten = 1234,3,Hangup()
I didn't test this code, but this is my tip the main idea is that you
need to catch de DID and make a GoTo for the context you want.
Best regards,
Marco Mouta
On 4/2/06, Rich Adamson [EMAIL PROTECTED] wrote:
Steve Gladden wrote:
What version of asterisk? (been lots of changes happening to the sip
code over the last year)
SVN-branch-1.2-r9156
Have you looked at the sample configs in /usr/src/asterisk/configs?
Yes I have and my own configs are pretty much copies of them.
They do not detail, do or explain the simple concept that I am
trying to accomplish.
If they do I don't see it.
#1 I have more than one incoming SIP account
#2 I would like to have them come into the context of
my choice when a call comes in.
HOW do I do this?
currently I have 3 register lines
there is no way to specify in a register line
some way of making the call start in any other context
other than what is specified in the [general] section
of sip.conf
It seems that somehow maybe if there is a peer tat is somehow
matched to the register line (how???) it may work.
There may be some crazy way to do this within a peer
if so this is the information I am looking for...
The examples and descriptions are not at all clear to me
I have 3 accounts with the same provider
How do I get incoming calls to come into three different contexts
that I will create is the question.
From the example file I see:
Asterisk can register as a SIP user agent to a SIP proxy (provider)
; Format for the register statement is:
; register = user[:secret[:[EMAIL PROTECTED]:port][/extension]
;
; If no extension is given, the 's' extension is used. The extension needs
to
; be defined in extensions.conf to be able to accept calls from this SIP
proxy
I actually need to do 3 of these.
;register = 2345:[EMAIL PROTECTED]/1234
;
;Register 2345 at sip provider 'sip_proxy'. Calls from this provider
;connect to local extension 1234 in extensions.conf, default context,
;unless you configure a [sip_proxy] section below, and configure a
;context.
Ok I have 3 accounts from the same provider
one [sip_proxy] section just puts me in the same problem boat I'm already
in using a register line
the calls some into the context specified in [general] section of sip.conf
I need to somehow differentiate the three SIP 'lines' and give
them different contexts to start in.
;Tip 1: Avoid assigning hostname to a sip.conf section like
[provider.com]
OK sure then how will this associate with my register line that
uses provider.com
This makes no sense to me...
I mean It really makes no sense...
Sorry for my confusion.
Do I need the register line or do I not need the register line?
Why even have a register line if you don't need it and can somehow
do this in a peerf, riend or user section.
and if you need the register line the instructions say
not to use [provider.com] as the peer, then how the heck do you
get that register line to work with an associated [peer].
I need to get a handle on how this works before I go posting my
sporatic attempts to get a friend,peer or user to 'register'
which is not working.
The only way I've been able to get my system to take incoming calls
from our sip provider so far is to use register lines and keep
the system 'registered' with our provider.
I don't use any sip providers, so be careful with what I say here.
Based on the current sip.conf.sample comments (as of today), it would
appear you need to do something like this:
register = 2345:[EMAIL PROTECTED]/1234
register = 2346:[EMAIL PROTECTED]/2345
register = 2347:[EMAIL PROTECTED]/3456
The above register statements are used to inform your sip provider which
IP address you are coming from, and calls for each of those three
accounts (eg, 2345, 2346, and 2347) will be routed to your system. In
your extensions.conf, you would need something like:
exten = 1234,1,Dial(SIP/3000)
exten = 2345,1,Dial(SIP/3001)
exten = 3456,1,Dial(SIP/3002)
Note the comments in the sample config relative to not using a host=
statement in the type=peer section. Also note the above register
statements assume the use of three different account names (eg, 2345,
2346, and 2347
RE: [Asterisk-Users] kernel recompilation on a asterisk server
Is it a 2.6 kernel? Did you included CRC_CCITT and RTC support when you made the make menuconfig? -Mensagem original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Em nome de nik600 Enviada: sábado, 1 de Abril de 2006 10:28 Para: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Assunto: Re: [Asterisk-Users] kernel recompilation on a asterisk server i've tried make clean make make install in zaptel...but i still get errors... particularry i get: wct4xxp: Unknown symbol zt_rbsbits wct4xxp: Unknown symbol zt_unregister wct4xxp: Unknown symbol zt_register wct4xxp: Unknown symbol zt_alarm_notify zaptel: Unknown symbol crc_ccitt_table zaptel: Unknown symbol crc_ccitt_table wct4xxp: Unknown symbol zt_ec_span wct4xxp: Unknown symbol zt_receive wct4xxp: Unknown symbol zt_qevent_lock wct4xxp: Unknown symbol zt_ec_chunk wct4xxp: Unknown symbol zt_transmit wct4xxp: Unknown symbol zt_rbsbits wct4xxp: Unknown symbol zt_unregister wct4xxp: Unknown symbol zt_register wct4xxp: Unknown symbol zt_alarm_notify zaptel: Unknown symbol crc_ccitt_table zaptel: Unknown symbol crc_ccitt_table wct4xxp: Unknown symbol zt_ec_span wct4xxp: Unknown symbol zt_receive wct4xxp: Unknown symbol zt_qevent_lock wct4xxp: Unknown symbol zt_ec_chunk wct4xxp: Unknown symbol zt_transmit wct4xxp: Unknown symbol zt_rbsbits wct4xxp: Unknown symbol zt_unregister wct4xxp: Unknown symbol zt_register wct4xxp: Unknown symbol zt_alarm_notify zaptel: Unknown symbol crc_ccitt_table zaptel: Unknown symbol crc_ccitt_table wct4xxp: Unknown symbol zt_ec_span wct4xxp: Unknown symbol zt_receive wct4xxp: Unknown symbol zt_qevent_lock wct4xxp: Unknown symbol zt_ec_chunk wct4xxp: Unknown symbol zt_transmit wct4xxp: Unknown symbol zt_rbsbits wct4xxp: Unknown symbol zt_unregister wct4xxp: Unknown symbol zt_register wct4xxp: Unknown symbol zt_alarm_notify zaptel: Unknown symbol crc_ccitt_table zaptel: Unknown symbol crc_ccitt_table wct4xxp: Unknown symbol zt_ec_span wct4xxp: Unknown symbol zt_receive wct4xxp: Unknown symbol zt_qevent_lock wct4xxp: Unknown symbol zt_ec_chunk wct4xxp: Unknown symbol zt_transmit wct4xxp: Unknown symbol zt_rbsbits wct4xxp: Unknown symbol zt_unregister wct4xxp: Unknown symbol zt_register wct4xxp: Unknown symbol zt_alarm_notify zaptel: Unknown symbol crc_ccitt_table zaptel: Unknown symbol crc_ccitt_table wct4xxp: Unknown symbol zt_ec_span wct4xxp: Unknown symbol zt_receive wct4xxp: Unknown symbol zt_qevent_lock wct4xxp: Unknown symbol zt_ec_chunk wct4xxp: Unknown symbol zt_transmit wct4xxp: Unknown symbol zt_rbsbits wct4xxp: Unknown symbol zt_unregister wct4xxp: Unknown symbol zt_register wct4xxp: Unknown symbol zt_alarm_notify zaptel: Unknown symbol crc_ccitt_table zaptel: Unknown symbol crc_ccitt_table wct4xxp: Unknown symbol zt_ec_span wct4xxp: Unknown symbol zt_receive wct4xxp: Unknown symbol zt_qevent_lock wct4xxp: Unknown symbol zt_ec_chunk wct4xxp: Unknown symbol zt_transmit wct4xxp: Unknown symbol zt_rbsbits wct4xxp: Unknown symbol zt_unregister wct4xxp: Unknown symbol zt_register wct4xxp: Unknown symbol zt_alarm_notify zaptel: Unknown symbol crc_ccitt_table zaptel: Unknown symbol crc_ccitt_table torisa: Unknown symbol zt_receive torisa: Unknown symbol zt_ec_chunk torisa: Unknown symbol zt_lboname torisa: Unknown symbol zt_transmit torisa: Unknown symbol zt_rbsbits torisa: Unknown symbol zt_unregister torisa: Unknown symbol zt_register torisa: Unknown symbol zt_alarm_notify where is 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 ___ --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] Sjphone looses registry in my Asterisk Server then i need to restart Sjphone PC, why?
Hi all, I've my Server running well, then sometimes Sjphones looses registry and it only works well again if i restart the pc running sjphone. Has any one experience this? Best regards, Marco Mouta ___ --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] Sjphone looses registry in my Asterisk Server then i need to restart Sjphone PC, why?
Windows XP service Pack 2 What you mean with SIP config look like? I've everything by default, only config for Calls through SIP proxy Bug patches from sjphone? On 3/30/06, Chuck Bunn [EMAIL PROTECTED] wrote: Hi, What does your SIP config look like for the SJPhone? Also what operating system does this PC have and is it up to date with security and bug patches. Thanks Marco Mouta wrote: Hi all, I've my Server running well, then sometimes Sjphones looses registry and it only works well again if i restart the pc running sjphone. Has any one experience this? Best regards, Marco Mouta ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SJphone Do not send silence - option ? Should be disabled for Asterisk
Hi all, I would like to hear from you, SjPhone has the option to Do not Send silence (audio options, advanced), should i use this or remove this option? Everything ran well until now, but there was few people on my server, i'm increasing sip extensions and i want to avoid complains from the users:) Best regards, Marco Mouta ___ --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] Dial out .call files File permissions??
Hi all, I've created this test.call file and it is not running outgoing call files: i've made mv test.call /var/spool/asterisk/outgoing and nothing happens Channel: SIP/200 MaxRetries: 3 RetryTime: 40 WaitTime: 25 Context: from-internal Extension: 200 Priority: 1 My asterisk is running with asterisk user. not root user. Could you help me on ? Could this be a problem of file permissions? ___ --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] Dial out .call files File permissions??
Thank you for your fast reply!!! It's working on for SIP:) I've tried to my zapata and doesn't make the call, i get this: Attempting call on ZAP/g1964391121 for [EMAIL PROTECTED]:1 (Retry 2) -- Attempting call on ZAP/[EMAIL PROTECTED] for [EMAIL PROTECTED]:1 (Retry 1) -- Attempting call on ZAP/[EMAIL PROTECTED] for [EMAIL PROTECTED]:1 (Retry 2) -- Attempting call on ZAP/[EMAIL PROTECTED] for [EMAIL PROTECTED]:1 (Retry 1) Do I need to define context to outbound calls through my ZAP ? Thanks in advance, Marco Mouta On 3/28/06, Tomislav Vojvodic [EMAIL PROTECTED] wrote: I copy/pasted Channel: SIP/200 MaxRetries: 3 RetryTime: 40 WaitTime: 25 Context: from-internal Extension: 200 Priority: 1 ..and saved it as 'callme' .. ..and put chmod 777 callme ..and mv callme /var/spool/asterisk/outgoing/ All as root - and.. it's working ;) (Tested on AAH 2.7) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Marco Mouta Sent: Tuesday, March 28, 2006 11:47 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Dial out .call files File permissions?? Hi all, I've created this test.call file and it is not running outgoing call files: i've made mv test.call /var/spool/asterisk/outgoing and nothing happens Channel: SIP/200 MaxRetries: 3 RetryTime: 40 WaitTime: 25 Context: from-internal Extension: 200 Priority: 1 My asterisk is running with asterisk user. not root user. Could you help me on ? Could this be a problem of file permissions? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dial out .call files File permissions??
it's working , the problem was: Channel: ZAP/g1X I changed to ZAP/g1/X And it's working fine! Thank you all On 3/28/06, Marco Mouta [EMAIL PROTECTED] wrote: Thank you for your fast reply!!! It's working on for SIP:) I've tried to my zapata and doesn't make the call, i get this: Attempting call on ZAP/g1964391121 for [EMAIL PROTECTED]:1 (Retry 2) -- Attempting call on ZAP/[EMAIL PROTECTED] for [EMAIL PROTECTED]:1 (Retry 1) -- Attempting call on ZAP/[EMAIL PROTECTED] for [EMAIL PROTECTED]:1 (Retry 2) -- Attempting call on ZAP/[EMAIL PROTECTED] for [EMAIL PROTECTED]:1 (Retry 1) Do I need to define context to outbound calls through my ZAP ? Thanks in advance, Marco Mouta On 3/28/06, Tomislav Vojvodic [EMAIL PROTECTED] wrote: I copy/pasted Channel: SIP/200 MaxRetries: 3 RetryTime: 40 WaitTime: 25 Context: from-internal Extension: 200 Priority: 1 ..and saved it as 'callme' .. ..and put chmod 777 callme ..and mv callme /var/spool/asterisk/outgoing/ All as root - and.. it's working ;) (Tested on AAH 2.7) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Marco Mouta Sent: Tuesday, March 28, 2006 11:47 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Dial out .call files File permissions?? Hi all, I've created this test.call file and it is not running outgoing call files: i've made mv test.call /var/spool/asterisk/outgoing and nothing happens Channel: SIP/200 MaxRetries: 3 RetryTime: 40 WaitTime: 25 Context: from-internal Extension: 200 Priority: 1 My asterisk is running with asterisk user. not root user. Could you help me on ? Could this be a problem of file permissions? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dial out .call files File permissions??
In fact i've never done it. And i don't have any Cisco Phone... If i find something i will report it here :) Best regards, Marco Mouta On 3/28/06, Tomislav Vojvodic [EMAIL PROTECTED] wrote: heh.. I just noticed that ;) Heh, do you know maybe how to update time/date on all Cisco 7905 phones through asterisk? I need to increase time for 1 hour.. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Marco Mouta Sent: Tuesday, March 28, 2006 12:37 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Dial out .call files File permissions?? it's working , the problem was: Channel: ZAP/g1X I changed to ZAP/g1/X And it's working fine! Thank you all On 3/28/06, Marco Mouta [EMAIL PROTECTED] wrote: Thank you for your fast reply!!! It's working on for SIP:) I've tried to my zapata and doesn't make the call, i get this: Attempting call on ZAP/g1964391121 for [EMAIL PROTECTED]:1 (Retry 2) -- Attempting call on ZAP/[EMAIL PROTECTED] for [EMAIL PROTECTED]:1 (Retry 1) -- Attempting call on ZAP/[EMAIL PROTECTED] for [EMAIL PROTECTED]:1 (Retry 2) -- Attempting call on ZAP/[EMAIL PROTECTED] for [EMAIL PROTECTED]:1 (Retry 1) Do I need to define context to outbound calls through my ZAP ? Thanks in advance, Marco Mouta On 3/28/06, Tomislav Vojvodic [EMAIL PROTECTED] wrote: I copy/pasted Channel: SIP/200 MaxRetries: 3 RetryTime: 40 WaitTime: 25 Context: from-internal Extension: 200 Priority: 1 ..and saved it as 'callme' .. ..and put chmod 777 callme ..and mv callme /var/spool/asterisk/outgoing/ All as root - and.. it's working ;) (Tested on AAH 2.7) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Marco Mouta Sent: Tuesday, March 28, 2006 11:47 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Dial out .call files File permissions?? Hi all, I've created this test.call file and it is not running outgoing call files: i've made mv test.call /var/spool/asterisk/outgoing and nothing happens Channel: SIP/200 MaxRetries: 3 RetryTime: 40 WaitTime: 25 Context: from-internal Extension: 200 Priority: 1 My asterisk is running with asterisk user. not root user. Could you help me on ? Could this be a problem of file permissions? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] * Meetme Freeze patch found
I'm a bit newbie, could you tell me how to i apply the patch? Thanks in advance Marco Mouta On 3/27/06, Benoit Panizzon [EMAIL PROTECTED] wrote: On Friday 24 March 2006 16:05, Benoit Panizzon wrote: Hi all Apparently there is a patch for those 1.2.4/5 MeetMe Freezes: http://bugs.digium.com/view.php?id=5884 Haven't tried it out yet. I can now confirm: No freezes/crashes anymore since I applied the patch. -Benoit- ___ --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] Phones were working fine - Now there is noaudiowhen calling between extensions
Are you using which version of Asterisk?? Did you check if you are facing the old audio bug on bridge calls that appeared ? http://asteriskvoip.blogspot.com/2006_01_01_asteriskvoip_archive.html Wednesday, January 25, 2006 Update: No audio - Update your Asterisk This morning we discovered a serious bug that stopped all bridged audio in our Asterisk servers. Mark found the problem and soon fixed it. If you get this problem today, please update your Asterisk server. A fix has been commited to the subversion repository for 1.2 as well as trunk. A fixed 1.2.3 release will be published on ftp.digium.com as soon as we can find a release engineer (consider the time zone problem). A big thank you to everyone in the IRC channel that helped us locate this issue and to Mark that fixed it so quickly. --- I hope it helps. Best regards, Marco Mouta On 3/22/06, Charles Marcus [EMAIL PROTECTED] wrote: C F wrote: Polycoms are not the best if you want a phone that works behind NAT. Do you mean in general? Or only if you are trying to interconnect multiple offices? Are Polycoms fine for just one office, if the entire office is behind a NAT device, and the phones are only being used for normal calling? Thanks, -- Charles ___ --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] I can't resume a call on hold from zap device
I have a strange problem: if I put on hold an incoming call from my Digium TE110P, I can't resume it and the person at the phone continues to hear MOH until the line falls. My TE110P is connected with an italian E1 NT. If I put on hold a call on a SIP channel I can resume it without any problems. Is there someone that can help me? These are my configurations: zaptel.conf span=1,1,0,ccs,hdb3,crc4 bchan=1-15 dchan=16 bchan=17-31 loadzone = it defaultzone=it zapata.conf: [trunkgroups] [channels] language=it signalling=pri_cpe switchtype=euroisdn usecallingpres=yes pridialplan=local prilocaldialplan=local nationalprefix=0 internationalprefix=00 faxdetect=both callwaiting=yes echocancel=yes immediate=no overlapdial=yes echocancelwhenbridged=yes echotraining=400 callerid=asreceived group=1 context=from-pstn channel = 1-15,17-31 Thanks in advance, Marco. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 10minutes to restart [EMAIL PROTECTED] 2.7
Hi all, I've bought a TE110P, and received it today. So i decided to install [EMAIL PROTECTED] 2.7 with this card. In the past i had experiencies with X100P (clone card) and it never take me so long to reboot the machine Machine: P4- 2,8Ghz 1GRAM TE110P What could be wrong? Best regards, Marco Mouta ___ --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] Directory doesn't work well [EMAIL PROTECTED] try from PSTN with Digital recepcionist- Directory based on Last name
Hi all,
Directory lookup, [EMAIL PROTECTED] 2.7, are this small bugs?
case DIR_FIRST: $intro = ($operator ? dir-intro-fn-oper :
dir-intro-fn); break;
case DIR_BOTH: $intro = ($operator ?
dir-intro-fnln-oper : dir-intro-fnln); break;
case DIR_LAST: default: $intro = ($operator ?
dir-intro-oper : dir-intro); break;
dir-intro-oper.gsm is not available on asterisk sound directory!
Also i have a doubt on Directory agi script, I found this:
else if (!empty($digits) || ($digits === 0)) {
// strict type checking as they may have
entered 0 (string) which is empty()
$agi-stream_file(dir-nomatch);
} // else, we timed out
Probably it's because i'm newbie, but is it correct 3 equals? ($digits
=== 0)) ?
Best regards,
Marco Mouta
___
--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] Development news :: T38 passthrough support
Hi, yes, i do believe that T38 on Asterisk is a huge step! I've red about propriatery solutions with it , i was just wondering when Asterisk would get it. This project is just perfect, every day keeping in track with every one needs. Tthank you for your excellent work Steve Underwood! Best regards, Marco Mouta On 3/10/06, Olle E Johansson [EMAIL PROTECTED] wrote: Friends in the Asterisk.org community, There is a lot of cool stuff going on in Asterisk development, things that will change Asterisk and make it work better in your organisation, make it easier to sell in your area or give you more consulting oppurtunities - in short, functionality that will make a lot of sense for you users. However, developers can't really get anywhere without a dialog with the users. You know what you need, you know what is missing and how you would like to make Asterisk a better choice. I am planning to send out a description of new features now and then, to inform you about what is going on, but also to get some feedback. The bug tracker is not only a tool for developers, but also for testers and users to react to changes and contribute. *** ITU T.38 -- Fax over VoIP Fax over VoIP is a hot issue. VoIP service providers encourage people to switch to VoIP, but often forget to mention that faxing over VoIP is like russian roulette. On a local LAN, it might work if you pick a clear channel codec like G.711. Steve Underwood, member of the Asterisk developer team, has writen a good article about the problems involved and the solutions for it on his web site, the URL is http://soft- switch.org/foip.html T.38 is an ITU standard for fax over VoIP. To simplify, the idea is to decode the fax audio stream at the ingress point, convert it to a data stream that is not sensitive towards jitter or delays and encode it into audio again if needed at the other end of the call - if you can't convert it to an image somewhere in the middle and print it directly, or send it by e-mail. *** T.38 PASSTHROUGH in Asterisk Steve is the main contributor behind the work for T38 support in Asterisk. He's also the author of spandsp - the fax application that many use in Asterisk. The first part is to be able to send T38 calls to your Asterisk PBX and make Asterisk recognize this and forward the data stream to another endpoint that supports T.38. Asterisk won't be an T.38 endpoint, but will handle T.38 calls properly, regardless if the T.38 was offered in the original call setup, or if the caller suddenly sends a fax in the middle of a call (a re-invite). The requirement is that the incoming channel and the outbound channel both supports T38. If not, the call will be declined in a proper way. When this is tested and stable, work will continue to see if we can make Asterisk an T.38 endpoint. This is a very important addition to Asterisk. There is code for testing available. If you are interested, please check this URL in the bug tracker: http://bugs.digium.com/view.php?id=5090 I think this is a big step for Asterisk. Do you? If so, don't forget to say thank you to Steve Underwood - Coppice! Have a nice weekend! /Olle --- * Olle E. Johansson - [EMAIL PROTECTED] * Asterisk Training http://edvina.net/training/ ___ --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] I don't listen first seconds of audio from call - Asterisk integration with old PBX
Hi all, i have: out side PSTNOldPBX-Analog-Asterisk (X100P) ^ | Local Ext What is happening is: Calls from Local ext goes to Asterisk and everything is fine. Calls from Out side PSTN reach the OldPBx and are bridged do Asterisk, but i don't listen firs audio seconds of autoattendant:( but after that every think works well). I've tried to put a Wait(6) on Asterisk after Answer() and it solved the problem... but i really wouldn't like to wait 6 seconds :( Any suggestions? Best regards, Marco Mouta ___ --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] Stress Tests from AsteriskGur with [EMAIL PROTECTED]
Hi all, I'm planning to test my two [EMAIL PROTECTED] one is 1.5 and another is 2.5 Does any one got already Astertest - asterisk stress testing tool working one? I've red Asterisk Guru, http://www.asteriskguru.com/tutorials/astertest.html and after all the tutorial still remaining questions from users with problems ( in fact i didn't find any sucessfull feedback). I'm a bit afraid of doing all the tutorial and get in troubles with my stable asterisks Any one has tried it? Best regards, Marco Mouta ___ --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] Receiving Multiple calls on asterisk at home
Could it be Call Waiting Deactived? On 3/7/06, Rolf Brusletto [EMAIL PROTECTED] wrote: All - I've been muddling around with this for a few days now.. and I'm trying to figure out why I am not receiving more than one phone call on each polycom 501 phone. I can make more than one phone call out, but not receive another one in, while on a call. Has anybody seen this behaivior before, or is there something simple in the config i'm missing, like.. maxcalls.. or something. Thanks! Rolf Brusletto ___ --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] Clock is runing too fast, [EMAIL PROTECTED] Ztdummy and VMware workstation
Hi all, I've [EMAIL PROTECTED] with Ztdummy running on VMWare, and i've adjust already three times the date and it seems to me it is running clock faster... After a while Asterisk clock greater than my windows clock time Isn't this strange? I'm just waiting for a Digium card to change this to a real Linux System. Does any one could help me understanding what is going on? Best regards, Marco Mouta ___ --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] Conference room owner Changing his room password? [EMAIL PROTECTED]
Hi all, I didn't find yet any info about this. Is there any way for a Conference Room Owner to change his own password? A kind of Menu like calling his conference room: example:8200 And an IVR option to change password. That seems to me interesting, because i may not want the same users entering two diferent days on my conference room... Also I don't think it is a good choice to contact Administrator to change my Meetme password. Best regards, Marco Mouta ___ --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] [EMAIL PROTECTED] Servers Connecting Portugal to Brazil (offices)
Hi all, I'm planning to connect 2 office from one company. I'm the developer, so i hope i can get all the features working well. [EMAIL PROTECTED](Portugal)-IAX2/[EMAIL PROTECTED](Brazil) 1- First i'm integrating Asterisk in Portugal's company office, one [EMAIL PROTECTED] with TE110P connecting to an old PBX. (the same is done in Brazil, but only VoIP no TE110P) For [EMAIL PROTECTED] PCs: -P4 1GRam 100GHard Disk (About 20 to 50 users initially) 2- For Portugal internal VoIP calls as well as VoIP to PSTN i think it would be all ok. I would like to hear from you? (using Alaw will I need QoS on our LAn? we have Gigabit Lan) Main doubts are the the connection between Brazil and Portugal. Will it work only using IAX2 and Alaw? Will I need G729 for this connections? Does DTMF works fine with G729? (I'm planning maximum 4 simultaneous calls to Brazil) We have broadband connection 4Mbit. I hope this Excellent mailing list could help me on giving me some Feedback and or advices/tips. Best regards, Marco Mouta ___ --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] nwebmail
Hi all, I got also your question, how to use nwebmail? Nwebmail is used for administration mail reports, i think. Take a look on this: http://www.vozdigital.org/modules.php?op=modloadname=Newsfile=articlesid=95 I've made login with user: admin password: mypassword_for_admin I'm developing a solution based on [EMAIL PROTECTED], and also trying to improve administration docs for my client, would be an extra value to understand what for nwebmail and main advantages... Basically it seems to be that the main cronjobs and main events are there on email messages, am I wrong? Best regards, Marco Mouta On 1/18/06, yrving rivas [EMAIL PROTECTED] wrote: Ok, thanks, it works for me. Regards, Yrving Dovid Bender [EMAIL PROTECTED] escribió: If you are new I would reccomend using [EMAIL PROTECTED] http://asteriskathome.soundforge.net . It is a great resource for beginers. Also get the book (again I dont have the URL if some one does please post it). Asterisk Regards, Dovid --- yrving rivas wrote: Hello! I am new to Asterisk, AMP, Linux...did I say all?.. I just installed Asterisk, and for my needs it is working great. In my AMP I see the nwebmail but I can´t get into it. When I place my login and password, comes with the following message: An internal error has occured. Please co ntact your system administrator. If you are the system administrator, check the log files. The log files don´t help me very much. Can someone tell me how to use the nwebmail?, how to get in for first time? Regards! Yrving - Do You Yahoo!? La mejor conexión a Internet y 2GB extra a tu correo por $100 al mes. http://net.yahoo.com.mx ___ --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 __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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 Do You Yahoo!? La mejor conexión a Internet y 2GB extra a tu correo por $100 al mes. http://net.yahoo.com.mx ___ --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] Destroying a SIP extension doesn't destroy voicemail box?is this a bug?
Hi all, I'm using [EMAIL PROTECTED] 2.5, and i've done: 1-Create a SIP extension. 2-Leave there a Voicemail message 3-Remove SIP extension Then I've create another SIP extension but with the same number of the above one. I found imediately a voicemail message in my voicemail box. Is this a bug? Am I doing something wrong? Best regards, Marco Mouta ___ --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] I can't receive multiple pages with spandsp
Hi all,
I'trying to use spandsp (app_rxfax) to receive faxes.
When there are more than one page, the system creates a tiff file with only the
first page and the other are lost, even if the full log says:
Mar 7 17:17:42 DEBUG[5876] app_rxfax.c:
==
Mar 7 17:17:42 DEBUG[5876] app_rxfax.c: Pages transferred: 1
Mar 7 17:17:42 DEBUG[5876] app_rxfax.c: Image size: 1728 x 1118
Mar 7 17:17:42 DEBUG[5876] app_rxfax.c: Image resolution7700 x 3850
Mar 7 17:17:42 DEBUG[5876] app_rxfax.c: Transfer Rate: 9600
Mar 7 17:17:42 DEBUG[5876] app_rxfax.c: Bad rows0
Mar 7 17:17:42 DEBUG[5876] app_rxfax.c: Longest bad row run 0
Mar 7 17:17:42 DEBUG[5876] app_rxfax.c: Compression type1
Mar 7 17:17:42 DEBUG[5876] app_rxfax.c: Image size (bytes) 0
Mar 7 17:17:42 DEBUG[5876] app_rxfax.c:
==
Mar 7 17:18:13 DEBUG[5876] app_rxfax.c:
==
Mar 7 17:18:13 DEBUG[5876] app_rxfax.c: Pages transferred: 2
Mar 7 17:18:13 DEBUG[5876] app_rxfax.c: Image size: 1728 x 1117
Mar 7 17:18:13 DEBUG[5876] app_rxfax.c: Image resolution7700 x 3850
Mar 7 17:18:13 DEBUG[5876] app_rxfax.c: Transfer Rate: 9600
Mar 7 17:18:13 DEBUG[5876] app_rxfax.c: Bad rows0
Mar 7 17:18:13 DEBUG[5876] app_rxfax.c: Longest bad row run 0
Mar 7 17:18:13 DEBUG[5876] app_rxfax.c: Compression type1
Mar 7 17:18:13 DEBUG[5876] app_rxfax.c: Image size (bytes) 0
Mar 7 17:18:13 DEBUG[5876] app_rxfax.c:
==
Mar 7 17:18:16 DEBUG[5876] app_rxfax.c:
==
Mar 7 17:18:16 DEBUG[5876] app_rxfax.c: Fax successfully received.
Mar 7 17:18:16 DEBUG[5876] app_rxfax.c: Remote station id: 3002
Mar 7 17:18:16 DEBUG[5876] app_rxfax.c: Local station id:
Mar 7 17:18:16 DEBUG[5876] app_rxfax.c: Pages transferred: 2
Mar 7 17:18:16 DEBUG[5876] app_rxfax.c: Image resolution: 7700 x 3850
Mar 7 17:18:16 DEBUG[5876] app_rxfax.c: Transfer Rate: 9600
Mar 7 17:18:16 DEBUG[5876] app_rxfax.c:
==
In extensions.conf I have:
exten = 1080,1,NoOp(Entro nel context from-FAX)
exten = 1080,2,Answer
exten = 1080,3,Macro(ricezionefax)
exten = 1080,4,system(tiff2ps -2 -a -e -z -w 8 -h 10.5 ${FAXFILE} | lpr [EMAIL
PROTECTED]) ;;;I send the fax to my printer
exten = 1080,5,Hangup
and
[macro-ricezionefax]
exten = s,1,SetVar(FAXFILE=/var/spool/asterisk/fax/${UNIQUEID}.tif)
exten = s,2,rxfax(${FAXFILE})
exten = s,102,Goto(2)
Is this a problem of spandsp (I'm using spandsp-0.0.2pre25)
or is there an error in my configuration?
Thanks in advance,
Marco.
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] BLF not working after reload
I solved that problem for Polycom phones with the patch at: http://bugs.digium.com/view.php?id=6047 ___ --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] BLF not working after reload
Hi Douglas, I'm using Asterisk-1.2.4 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Polycom IP 601 Buddy Watch doesn't work after Asterisk reload
Hi, I configured Buddy Watch function on my Polycom IP 601. It works well, until I make a reload of Asterisk. After reload, if I give the show hints command in Asterisk's CLI, it says that there are no watcher for the extensions that I configured. Before the reload in the CLI appears: -= Registered Asterisk Dial Plan Hints =- 3002 : SIP/3002 State:Idle Watchers 1 3006 : SIP/3006 State:Idle Watchers 1 3003 : SIP/3003 State:Unavailable Watchers 1 3001 : SIP/3001 State:Idle Watchers 1 3000 : SIP/3000 State:Idle Watchers 1 After the reload in the CLI appears: -= Registered Asterisk Dial Plan Hints =- 3002 : SIP/3002 State:Idle Watchers 0 3006 : SIP/3006 State:Idle Watchers 0 3003 : SIP/3003 State:Unavailable Watchers 0 3001 : SIP/3001 State:Idle Watchers 0 3000 : SIP/3000 State:Idle Watchers 0 Asterisk sends a SIP NOTIFY message in which the field Subscription-State is: terminated; reason=probation and the phone responds with a ACK. I have then to restart the phone to reactivate the Buddy Watch function. Is there anybody that can help me with this problem? Is it a problem of the PBX or a problem of the phone? Thanks in advance, regards, Marco. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Polycom IP 601 Buddy Watch problems
Hi, I configured Buddy Watch function on my Polycom IP 601. It works well, until I make a reload of Asterisk. After reload, It can't monitor any lines and I have to restart the phone to reactivate this function. Is this a specific problem of asterisk-1.2.3? How can I solve it? Thank in advance, regards, Marco. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FaxToEmail for diferent Channels and different Mail accounts?
Hi all, I'm going to buy E1 digium110P ,any one knows how i can get faxtomail working for three different channels? I mean: channel1--[EMAIL PROTECTED] channel2--[EMAIL PROTECTED] for 1 channel is not dificult [EMAIL PROTECTED] and NVfaxdetect Using [EMAIL PROTECTED] 2.5 faxToPDFmail works, but i always get error opening pdf in outlook. i've solved this with a mimeconstruct update explained on wiki for [EMAIL PROTECTED] 1.5, Does any one knows why [EMAIL PROTECTED] 2.5 the problem stills happening... It seems to me i will need to do the same...??? Best Regards, MoutaPT ___ --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 running on DMZ (no NAT) PROBLEMS- OPTION message is out of State
Hello, Currenly I've [EMAIL PROTECTED] 1.5 running on DMZ. I can register SJphone there, good audio on 8200 (webmeet me calls) and i also can dial Zapata extensions. When I dial sip phone extensions nothing happens if the client that i'm calling is registred, if the client has voicemail it goes to voicemail. IMPORTANT: I get this error message on my Check Point Firewall: sip reason:Attack Info - Malformed SIP datagram, OPTION message is out of State By the way i've one client that is running all ok, the others all have this problem. I hope some one could help me with this. Best regards, Marco Mouta ___ --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] which ATA SIP is better with asterisk
Hi i'm developing a solution with ASterisk, but in fact i don't know which ATA SIP device should buy. Could you give me some advices? Marco Mouta ___ --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] Fwd: Which ATA device do you recommend?
-- Forwarded message -- From: Marco Mouta [EMAIL PROTECTED] Date: Feb 15, 2006 1:58 PM Subject: Which ATA device do you recommend? To: [EMAIL PROTECTED] Hello, I'm developing a Voip Solution for a client, which ATA SIP do you recommend? there are some ATA devices fully tested with Asterisk? I hope that Asterisk experient users could give me their advice based on their experiencies. Thanks to all, ___ --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 running on DMZ (no NAT) PROBLEMS- OPTION message is out of State
Hello, Currenly I've [EMAIL PROTECTED] 1.5 running on DMZ. I can register SJphone there, good audio on 8200 (webmeet me calls) and i also can dial Zapata extensions. When I dial sip phone extensions nothing happens if the client that i'm calling is registred, if the client has voicemail it goes to voicemail. IMPORTANT: I get this error message on my Check Point Firewall: sip reason:Attack Info - Malformed SIP datagram, OPTION message is out of State By the way i've one client that is running all ok, the others all have this problem. I hope some one could help me with this. Best regards, Marco Mouta ___ --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: X100P help required
It seems to me, that your problem is that X100P is not detecting that the caller has hangup through PSTN. I really got lots of problems with disconnect detection, and currently i only get it working on asterisk @home 1.5 , it doesn't work well on later releases. The main changes i've made in zapata.conf are: busydetect=yes busypattern=500,500 busycount=6 callprogress=yes I'm not sure if the problem is that my X100P isn't from Digium, it's from X100P.com. Also i'm not an asterisk GURU. Hope this helps. Ps. If someone nows why it only works well on [EMAIL PROTECTED] please tellme. On 2/8/06, Tejas Shah [EMAIL PROTECTED] wrote: hello all, I m joined this asterisk group few months back. Actually i have installed asterisk on my PC and using X100P PSTN interface card to make this PC work as a single line VoIP gateway. Now i have one problem which is as follows: As i told i m using X100P card. I am getting good quality of speech from it whenever i m making call or receiving call through it. Now I have installed Soft X-Lite ip phone on 3 pc's. Now whenever i make call from analog phone to any IP phone, we can talk. After finishing talk I hang up the analog phone. Now when just after 2-3 mins when i want to make call to any IP phone, from my analog phone i m getting engaged tone. I have to restart my asterisk server, to come out of that engaged tone . Now evertime i cant restart asterisk server. I usually uses following commands to start asterisk server each time: first safe_asterisk then asterisk -r. So i m confused now, why this is happening? 1) Am i starting asterisk server in wrong way? or 2) Is there any problem with X100P card? Where i m wrong, i m not getting? hope all of u will help me to solve this problem. Thanks Tejas Relax. Yahoo! Mail virus scanning helps detect nasty viruses! ___ --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 LDAP Authentication Problem
did you notice the two dots in the IP address of ldaphost ? Marco. Chandan Mishra wrote: Hi I want to authenticate the asterisk users from the LDAP directory server not from the sip.conf. I tried to use the astirectory-1.2 http://www..asterisk-ev.org/download/astirectory-1.2-0.3.tgz . But i am not able to configure it properly. If somebody used it then please help. In the res_ldap.conf file i made the following entries. I am using my normal username and password to connect my asterisk server to the LDAP server of my organization. Is some administrator login is requried to connect ? [general] ldapuser=cn=chandan.mishra,dc=synapse,dc=com ldaphost=ldap://192.168.0..16 ldap://192.168.0.16 ldappass=chandan123 ldapbasedn=dc=synapse,dc=com After this the asterisk is not able to connect to the ldap database. And hence asterisk is not able to start. Its giving me following errors: == Parsing '/etc/asterisk/res_ldap.conf': Found Jan 17 23:38:09 WARNING[11207]: res_config_ldap.c:615 parse_config: LDAP RealTime: No database host found, using localhost via socket. Jan 17 23:38:09 WARNING[11207]: res_config_ldap.c:630 parse_config: LDAP RealTime Host: ldap://localhost Jan 17 23:38:09 WARNING[11207]: res_config_ldap.c:631 parse_config: LDAP RealTime User: cn= chandan.mishra,dc=synapse,dc=com Jan 17 23:38:09 WARNING[11207]: res_config_ldap.c:632 parse_config: LDAP RealTime Base DN: dc=synapse,dc=com Jan 17 23:38:09 ERROR[11207]: res_config_ldap.c:708 ldap_reconnect: LDAP failed to bind (host= ldap://localhost, user=cn=chandan.mishra,dc=synapse,dc=com Can't contact LDAP server 0)! Jan 17 23:38:09 ERROR[11207]: res_config_ldap.c:708 ldap_reconnect: LDAP failed to bind (host= ldap://localhost, user=cn=chandan.mishra,dc=synapse,dc=com Can't contact LDAP server 1)! Jan 17 23:38:09 ERROR[11207]: res_config_ldap.c:708 ldap_reconnect: LDAP failed to bind (host=ldap://localhost ldap://localhost, user=cn=chandan.mishra,dc=synapse,dc=com Can't contact LDAP server 2)! Jan 17 23:38:09 WARNING[11207]: res_config_ldap.c:539 load_module: LDAP RealTime: Couldn't establish connection. Check debug. Thanks Chandan ___ --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] CallProgress breaks DTMF
Hi, I enabled Callprogress in the zapata.conf , so in the CDR it will log other things other then answered (Busy, no answer etc), but, this seems to break my Polycom's DTMF, i configured RFC2833 for the dtmf in the sip.conf, and when callprogress is enabled, the dtmf doesnt reach the other end, Any idea/solution ? Thanks. Marco. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CallProgress breaks DTMF - RFC2833
Hi I enabled Callprogress in the zapata.conf , so in the CDR it will log other things other then answered (Busy, no answer etc), but, this seems to break my Polycom's DTMF, i configured RFC2833 for the dtmf in the sip.conf, and when callprogress is enabled, the dtmf doesnt reach the other end, Any idea/solution ? Thanks. Marco. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Hangup detection - TDM400P
Hi, I have a long delay when detecting hangups on the TDM400P card, with 4 FXO ports, When an incoming call dial's in, when hanging up, the asterisk will detect the hangup only after 10 seconds, i searched around, and found many similar problems, but no solution, i tried some options in zapate.conf , but nothing helped, any solution ? the lines are coming from SBC in San Fransisco, i asked them if i have disconnect supervision, and they said i do have it. Marco. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CallerID Length
Hi, I have a problem with the Caller ID string, seems like asterisk will display only 10 digits of the caller id. If the string is longer then 10 digits, asterisk will sometimes strip the first digit, and sometimes the last digits, in order to show a 10-digit callerid, Is this configurable ? i would like to get the caller id of international callers , with all digits. Any solution ? Thanks. Marco. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hangup detection - TDM400P
Yes, didnt change anything Marco. Angelito Manansala wrote: hmmm di you try this ;hanguponpolarityswitch=yes Cheerz! On 11/17/05, Marco Supino [EMAIL PROTECTED] wrote: Hi, I have a long delay when detecting hangups on the TDM400P card, with 4 FXO ports, When an incoming call dial's in, when hanging up, the asterisk will detect the hangup only after 10 seconds, i searched around, and found many similar problems, but no solution, i tried some options in zapate.conf , but nothing helped, any solution ? the lines are coming from SBC in San Fransisco, i asked them if i have disconnect supervision, and they said i do have it. Marco. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards, Angelito Manansala www.voicefidelity.net Mobile: +639175425807 DID: (+63) 44 7906770 msn: [EMAIL PROTECTED] skype: bulcrack ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] PRI pass-through
Hi, I want to build a PRI pass-through with a Cisco 2600, with two VWIC E1 cards, is this possible ? and do i need any other modules except for the E1 modules ? What i want to do is connect the asterisk to the PRI through the Cisco router, and let my legacy PBX utilize some of the PRI channels while testing Asterisk, Anyone with experience, sample configs or idea, please contribute. Thanks. Marco. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Detect registered peers
Hi, Is there a way to detect (in the dialplan) if a SIP peer is registered with the server ? I am using macros to dial to extension, becuase i dont want to define each extension in the dialplan, and, for example, my numbers are 8xx , i want to know if a peer exists/registered before ringing the line, i need something like Voicemailexists , but for SIP peers. any solution ? Thanks. Marco. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] fxotune fails with valid TDM/FXO card
Hi, I am using Asterisk 1.0.9 with the 1.2.0 zaptel, just for the fxotune utility, which solved my echo problems , my zttest results are low, but no echo on ZAP lines... Marco. Chris Miller wrote: Mojo with Horan Company, LLC wrote: The recent suggestion on the list was to not use 1.0.9 zaptel You mean the driver, or the version of fxotune? fxotune has been removed from the prior versions of the zaptel driver, it's only included in 1.2 now. As for the driver, is anyone using the 1.2 zaptel driver with Asterisk 1.0.9? The way the downloads are grouped together on the Asterisk web page, I was led to believe they shouldn't be mixed. Chris ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
R: [Asterisk-Users] Astricon - materials
Some people is still waiting for last Astricon materials; what about them ? Regards. Marco Vescovi -Messaggio originale- Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Per conto di Olle E. Johansson Inviato: mercoledì 26 ottobre 2005 8.42 A: Asterisk Users Mailing List - Non-Commercial Discussion Oggetto: Re: [Asterisk-Users] Astricon - materials marek cervenka wrote: hi, will be somewhere materials (videos, presentations) from astricon? Registered attendees will get information about the material soon. No videos where recorded this year. The 1.2 presentation I made together with Kevin has been available for a while at http://www.astricon.net/asterisk1-2/ and will be updated soon. Regards /Olle ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.361 / Virus Database: 267.12.5/149 - Release Date: 25/10/2005 * Questa e-mail, ed i suoi eventuali allegati, contengono informazioni confidenziali e riservate. Se avete ricevuto questa comunicazione per errore non utilizzatene il contenuto e non portatelo a conoscenza di alcuno. Siete inoltre pregati di elimi- narla dalla vostra casella e avvisare il mittente. E' da rilevare inoltre che l'attuale infrastruttura tecnologica non puo' garantire l'autenticita' del mittente, ne' tanto- meno l'integrita' dei contenuti.Opinioni, conclusioni ed altre informazioni contenu- te nel messaggio possono rappresentare punti di vista personali a meno di diversa esplicita indicazione autorizzata. * * Questa e-mail, ed i suoi eventuali allegati, contengono informazioni confidenziali e riservate. Se avete ricevuto questa comunicazione per errore non utilizzatene il contenuto e non portatelo a conoscenza di alcuno. Siete inoltre pregati di elimi- narla dalla vostra casella e avvisare il mittente. E' da rilevare inoltre che l'attuale infrastruttura tecnologica non puo' garantire l'autenticita' del mittente, ne' tanto- meno l'integrita' dei contenuti.Opinioni, conclusioni ed altre informazioni contenu- te nel messaggio possono rappresentare punti di vista personali a meno di diversa esplicita indicazione autorizzata. * * Questa e-mail, ed i suoi eventuali allegati, contengono informazioni confidenziali e riservate. Se avete ricevuto questa comunicazione per errore non utilizzatene il contenuto e non portatelo a conoscenza di alcuno. Siete inoltre pregati di elimi- narla dalla vostra casella e avvisare il mittente. E' da rilevare inoltre che l'attuale infrastruttura tecnologica non puo' garantire l'autenticita' del mittente, ne' tanto- meno l'integrita' dei contenuti.Opinioni, conclusioni ed altre informazioni contenu- te nel messaggio possono rappresentare punti di vista personali a meno di diversa esplicita indicazione autorizzata. * ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] App_directory + Festival
Hi, As anyone tried integrating App_Directory with any Text2Speech mechanism like festival ? Marco. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RealTime problem with sipusers accounts
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 I need to add and remove Sip accounts in realtime. What's the best way at the moment to do that? * Add/remove the user into the sip.conf and execute asterisk -x 'sip reload' ? Thanks for help Marco Kevin P. Fleming schrieb: Marco Balmer wrote: Server1 acts as a SIP Client only. Server2 should act as a SIP-Server with the sip_buddies table on the MySQL-Server. But this is not currently implemented. There is a patch in the bug tracker that will help move in this direction, but it's only a start, there are many more issues that need to be resolved for this to work properly. -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.1 (GNU/Linux) Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org iD8DBQFDUpHq8JLvhlgYtaoRAqOEAKCXsI3TLL23DDpzzMZi3cno4xqOTQCfUzX2 GCaR660+WeEHV/HayHwm4qY= =Sm3A -END PGP SIGNATURE- ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RealTime problem with sipusers accounts
Hello On Fri, 14 Oct 2005 01:25:20 -0500, Kevin P. Fleming wrote Marco Balmer wrote: Any ideas or hints? Yes. Whatever documentation told you that you could share a Realtime SIP peer database between two Asterisk servers was in error (or at least very incomplete). Server1 acts as a SIP Client only. Server2 should act as a SIP-Server with the sip_buddies table on the MySQL-Server. Thanks Marco ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Broadvoice Outages?
Yes, i am having timeouts on registering to the LAX sip server of broadvoice. Marco. Nate Kapi wrote: I've been having a lot of problems with Broadvoice lately. Anyone else been without service for extended periods of time this week? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RealTime problem with sipusers accounts
Hello @ all,
I hope you can help me.
server1: asterisk-cvs HEAD 2005-10-13
server2: asterisk-cvs HEAD 2005-10-13
I've configured RealTime (sipusers) on server2 together with a MySQL database.
The account in the database exists. It seems to be configured right. Then I
can read realtime infos with commands like realtime load sipusers name 301
But Server1 doesn't find the configured accounts.
server1:
Oct 14 06:56:13 WARNING[8523]: chan_sip.c:9507 handle_response_register: Got
404 Not found on SIP register to service [EMAIL PROTECTED], giving up
Any ideas or hints?
Thank you for help
Marco
server2*CLI sip show users
Username Secret Accountcode Def.Context
ACL NAT
server2*CLI realtime load sipusers name 301
Column Name Column Value
id 6
name 301
callerid 301
canreinvite yes
context cmo-incoming
fromuser 301
nat no
snip
/etc/asterisk/extconfig.conf
[settings]
sipusers = mysql,asterisk_db,sip_buddies
/etc/asterisk/res_mysql.conf
[general]
dbhost = localhost
dbname = asterisk_db
dbuser = asterisk
dbpass = xxx
dbport = 3306
dbsock = /var/run/mysqld/mysqld.sock
--
PGP Key - http://www.micressor.ch/GPG/gpg-key.txt
http://web.swissjabber.ch - xmpp/jabber: [EMAIL PROTECTED]
VoIP - sip:[EMAIL PROTECTED]
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] zttest - 100% ?
Hi, I would like to know what type of configuration could get me closer to 100% hits in zttest, when testing a TDM400P with 4 FXO ports, I am currently running kernel 2.4.31, on a IBM Xseries 306, with 3gh CPU, HT is disabled, PCI latency was changed, i still cant get more then 99.975% in the zttest testings, Thanks for any info. Marco. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] zttest - 100% ?
Hi, My TDM is on its own IRQ, and the x306 has only one full-size PCI slot.. so no playing with it, what results do you get from zttest ? what IRQ is the card on ? Marco. Damian Funnell wrote: Have you checked that the TDM400P isn't sharing an IRQ with anything else? Don't trust /proc/interrupts - run lspci -v to confirm this. We have * running on an x206 and found that the only way to stop the TDP400P sharing an IRQ with other devices was to juggle cards between slots. Hope this helps! Damian. FFF Managed Technology Ltd 60 Cook St P.O. 6368 Wellesley St Auckland t +64 9 356 2911 f +64 9 358 9070 m +64 21 415 297 w www.fff.co.nz Marco Supino wrote: Hi, I would like to know what type of configuration could get me closer to 100% hits in zttest, when testing a TDM400P with 4 FXO ports, I am currently running kernel 2.4.31, on a IBM Xseries 306, with 3gh CPU, HT is disabled, PCI latency was changed, i still cant get more then 99.975% in the zttest testings, Thanks for any info. Marco. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IBM x306 - some progress
Hi, I asked yesterday about a problem with x306 and IRQ sharing, didnt get much info, now, i was playing with lspci, and see something strange, lspci -v shows me the TDM400P card is on IRQ 7, and the SCSI card is also on IRQ 7, lspci -bv (from the man - b - shows bus-centric view, as seen by the BUS and not by the kernel) shows me the TDM400P is on IRQ 5, why does the kernel puts it on IRQ 7 ? any insights much appriciated. Marco. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IBM x306
Hi, This is a little off-topic,but if someone has any info, it could help me a LOT!, I am building an asterisk pbx (1.0.9) on an IBM x306 SCSI machine,my problem is that the BIOS assigns the same IRQ to the SCSI controller, and the TDM400P, i have tried several options of making the bios change the IRQ, but it will always move them together, anyone with some info about my options ? Thanks, Marco. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IBM x306
Only one PCI slot can hold the full size card like the TDM400P , the other slot has a smaller opening on the case. Marco. Alexander Lopez wrote: Can you try a different slot on the PCI bus?? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Marco Supino Sent: Saturday, September 24, 2005 8:51 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] IBM x306 Hi, This is a little off-topic,but if someone has any info, it could help me a LOT!, I am building an asterisk pbx (1.0.9) on an IBM x306 SCSI machine,my problem is that the BIOS assigns the same IRQ to the SCSI controller, and the TDM400P, i have tried several options of making the bios change the IRQ, but it will always move them together, anyone with some info about my options ? Thanks, Marco. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IBM x306
Hi, I tried setpci INTERRUPT_LEVEL (or something similar, cant remmeber now), and also setpci seems like it changed the IRQ, lspci -v still shows the old IRQ Marco. Stefan de Konink wrote: On Sun, 25 Sep 2005, Marco Supino wrote: I am building an asterisk pbx (1.0.9) on an IBM x306 SCSI machine,my problem is that the BIOS assigns the same IRQ to the SCSI controller, and the TDM400P, i have tried several options of making the bios change the IRQ, but it will always move them together, anyone with some info about my options ? Linux usually don't care about Bios settings, you could try kernel cmdline parameters. Acpi and IRQ are google terms for it. Stefan ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SpanDSP - Squished Faxes
Richard Cook ha scritto: Hello, Has anyone had issues with faxes showing up squished in the TIFF file? Any ideas what could be causing it? there's a faq on the spandsp site. the problem is not with spandsp. it's with the image visualization program. (i.e. irfanview 3.97 (win32) has the bug, i've contacted the author and he has fixed it and the fix, hopefully, will be included in the next release.) ciao ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Libtiff 3.5.7 - recommended version for spandsp
Roger Schreiter ha scritto: Marco Parmeggiani wrote: ... i had no problems receiving faxes with version 3.7.2. on the other hand i have big problems in sending multipage faxes. only Hi, where did you get that version? On libtiff.org, 3.6.1 is the most recent one. you're pointing to the wrong page: http://www.remotesensing.org/libtiff/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] communication between IAX softphones
I tried with several iax softphones: iaxcomm idefix iaxphone and i have a problems that i do not have with SIP clients. A calls B, B phone starts ringing, asterisk says that call has been accepted, that is ringing but it is not yet answered. If B picks up, asterisk says that call has been answered but, *before* User B pick up, he is already able to hear User A and viceversa. ciao ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Libtiff 3.5.7 - recommended version for spandsp
Roger Schreiter ha scritto: Hi, package tiff-v3.5.7 contains the currently recommended version of libtiff in order to run spandsp (fax support for asterisk). i had no problems receiving faxes with version 3.7.2. on the other hand i have big problems in sending multipage faxes. only the first page goes through. i've downloaded 3.5.7 but transmission problems remain. ciao ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] call file ignored?
Remco Barende ha scritto: Do you see anything on the console even if you dial a number that isn't answered? i see this for a non existant number: Attempting call on Zap/g1/12345 for [EMAIL PROTECTED]:1 (Retry 1) i guess it prints out for every call originated by a call file. asterisk -cvvv ciao ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Libtiff 3.5.7 - recommended version for spandsp
Marco Parmeggiani ha scritto: on the other hand i have big problems in sending multipage faxes. only the first page goes through. uhm, no, neither the first page is received. i was optimistic. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TxFax: can't get a fax to destination (log inside)
Can someone explain me what's going on and why the receiver of this fax guives up saying communication error? Slow carrier up Slow carrier down Slow carrier up CSI: 40 20 20 20 20 20 20 20 34 39 34 35 36 34 39 35 30 20 39 33 2b CSI without final frame tag Remote fax gave CSI as: +39 059465494 DIS: 80 00 ee f8 c4 00 DIS with final frame tag In state 10 DIS: Prefer 256 octet blocks Can receive fax Supported data signalling rates: V.27ter, V.29 and V.17 R8x7.7lines/mm and/or 200x200pels/25.4mm 2D coding Scan line length: 215mm Recording length: A4 (297mm) and B4 (364mm) Receiver's minimum scan line time: 0ms at 3.85 l/mm: T7.7 = T3.85 Error correction mode T.6 coding DCS: Can receive fax Selected data signalling rate: V.29, 9600bps 2D coding Scan line length: 215mm Recording length: A4 (297mm) Minimum scan line time: 20ms Minimum scan line time for higher resolutions: T15.4 = T7.7 Start sending document Start tx document Changed from phase 2 to 4 DCS: 83 00 86 80 80 80 00 HDLC underflow in state 3 Changed from phase 4 to 6 Changed from phase 6 to 3 Slow carrier up CFR: 84 CFR with final frame tag In state 4 Trainability test succeeded Start tx page 0 Slow carrier down Changed from phase 3 to 6 Changed from phase 6 to 4 Start tx page 1 EOP: 2f HDLC underflow in state 14 Changed from phase 4 to 3 Slow carrier up RTN: 4c RTN with final frame tag In state 14 Changed from phase 3 to 4 DCN: fb HDLC underflow in state 2 Disconnecting Changed from phase 4 to 7 Changed from phase 7 to 8 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] auto-dial dial status
I'm using autodial in conjuction with TxFax to send faxes on demand. An home made application generates the call file and puts it in the outgoing spool, the file is like this: Channel:Zap/g1/1232314324 MaxRetries:0 RetryTime:60 WaitTime:20 Context:faxout Extension:s SetVar:FAX_FILE=/shared/awfax/test.tif the extension called is this: [faxout] exten = s,1,TxFax(/shared/awfax/test.tif|caller) exten = s,2,Hangup My problem is that if asterisk can't connect to the called end then it doesn't go to the extension, so i am unable to report the error if the called end does not respond, does not exist or refuse the call. Is there some trick (or an elegant solution as well) to solve this problem? ciao ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] bristuff-0.2.0-RC8g: zaptel error in suse 9.2
Manuel Casal ha scritto: I made the make menuconfig and make dep in the kernel sources. i do not remember well how i solved that problem but i'm sure that make dep will issue you a warning and stop. run make to start the kernel build process and then stop it after few seconds. it will create the necessary symlinks in the kernel tree. maybe there's a more elegant solution but this should work. ciao ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] bristuff-0.2.0-RC8g: zaptel error in suse 9.2
Manuel Casal ha scritto: make[1]: Entering directory `/usr/src/linux-2.6.8-24.16-obj/i386/smp' make[1]: *** No rule to make target `modules'. Stop. make[1]: Leaving directory `/usr/src/linux-2.6.8-24.16-obj/i386/smp' make: *** [linux26] Error 2 linux:/usr/src/asterisk/bristuff-0.2.0-RC8g/zaptel-1.0.7 # Now what?:( i'm using a Debian. i'm missing those *-obj links in my /usr/src drwxr-xr-x 19 root root 4096 Jun 17 18:04 kernel-source-2.6.11 lrwxrwxrwx 1 root src18 May 24 14:36 linux - /usr/src/linux-2.6 lrwxrwxrwx 1 root src20 May 24 13:54 linux-2.6 - kernel-source-2.6.11 and it compiles fine. HTH ciao ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] unamble to dialout to mobiles and others special numbers
Asterisk 1.0.7-BRIstuffed-0.2.0-RC8a on a Debian 3.1 The system is connected with an HFC card directly to the telco line card is in TE mode and signalling used is bri_cpe_ptmp I am able to dial out some numbers and some not. In particular it seems that i can't call mobiles and special telco numbers like the information call center, emergency numbers,... If i use a normal hardware isdn phone i am able to do such calls. This is a call that works: -- Executing NoOp(SIP/11-1ecc, Call to 756756756) in new stack -- Executing GotoIf(SIP/11-1ecc, 0?3:5) in new stack -- Goto (default,059305698,5) -- Executing GotoIf(SIP/11-1ecc, 0?6:8) in new stack -- Goto (default,059305698,8) -- Executing NoOp(SIP/11-1ecc, External call) in new stack -- Executing Goto(SIP/11-1ecc, esterni|756756756|1) in new stack -- Goto (esterni,059305698,1) -- Executing Dial(SIP/11-1ecc, Zap/g1/756756756) in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called g1/756756756 -- Zap/1-1 is ringing [now i hangup] -- Hungup 'Zap/1-1' == Spawn extension (esterni, 756756756, 1) exited non-zero on 'SIP/11-1ecc' -- Executing Goto(SIP/11-1ecc, default|h|1) in new stack -- Goto (default,h,1) -- Executing Hangup(SIP/11-1ecc, ) in new stack == Spawn extension (default, h, 1) exited non-zero on 'SIP/11-1ecc' == Primary D-Channel on span 1 down == Primary D-Channel on span 1 up This is a call that does NOT work (ir. i'm calling my mobile phone): == Primary D-Channel on span 1 down == Primary D-Channel on span 1 up -- Executing NoOp(SIP/11-9d74, Call to 3777) in new stack -- Executing GotoIf(SIP/11-9d74, 0?3:5) in new stack -- Goto (default,3777,5) -- Executing GotoIf(SIP/11-9d74, 0?6:8) in new stack -- Goto (default,3473042866,8) -- Executing NoOp(SIP/11-9d74, External call) in new stack -- Executing Goto(SIP/11-9d74, esterni|3777|1) in new stack -- Goto (esterni,3777,1) -- Executing Dial(SIP/11-9d74, Zap/g1/3777) in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called g1/3777 -- Channel 0/1, span 1 got hangup Jun 16 13:07:17 WARNING[17330]: app_dial.c:412 wait_for_answer: Unable to forward voice Jun 16 13:07:17 WARNING[17330]: app_dial.c:412 wait_for_answer: Unable to forward voice -- Hungup 'Zap/1-1' == No one is available to answer at this time -- Executing Answer(SIP/11-9d74, ) in new stack -- Executing Playtones(SIP/11-9d74, congestion) in new stack -- Executing Congestion(SIP/11-9d74, ) in new stack Some configuration files: http://marcopar.altervista.org/extensions.conf http://marcopar.altervista.org/zapata.conf http://marcopar.altervista.org/zaptel.conf in the system messages i'm getting this: Zapata Telephony Interface Registered on major 196 PCI: Enabling device :00:06.0 ( - 0003) ACPI: PCI interrupt :00:06.0[A] - GSI 17 (level, low) - IRQ 185 zaphfc: CCD/Billion/Asuscom 2BD0 configured at mem 0xd08eaf00 fifo 0xcf338000(0xf338000) IRQ 185 HZ 1000 zaphfc: Card 0 configured for TE mode zaphfc: 1 hfc-pci card(s) in this box. Zaptel Configuration == SPAN 1: CCS/ AMI Build-out: 399-533 feet (DSX-1) Channel map: Channel 01: Individual Clear channel (Default) (Slaves: 01) Channel 02: Individual Clear channel (Default) (Slaves: 02) Channel 03: D-channel (Default) (Slaves: 03) 3 channels configured. frequently i get: zaphfc: empty HDLC frame or bad CRC received (framelen = 6, stat = 0xff, card = 0). ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] unamble to dialout to mobiles and others special numbers
Matteo Brancaleoni ha scritto: I am able to dial out some numbers and some not. In particular it seems that i can't call mobiles and special telco numbers like the information call center, emergency numbers,... try with: pridialplan=unknown prilocaldialplan=unknown it works. thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] zaphfc: empty HDLC frame or bad CRC received
Nick Barnes ha scritto: I've only ever seen when the signalling is wrong. For example if the line is in PTMP mode when it should be in PTP or vice-versa. this is the zapata.conf: group = 1 context=default signalling = bri_net_ptmp channel = 1-2 So, you're using NT mode PTMP signalling. Is the Asterisk box plugging into an ISDN circuit provided by a telco? If it is, then use bri_cpe_ptmp (for Point to MultiPoint) or bri_cpe (for Point to Point) instead of bri_net_ptmp. If it's plugged into a different ISDN device and needs to be in NT mode, then try bri_net instead. you pointed me in the right direction. the card is connected directly to the telco isdn and it should run in TE mode, also the signalling should be bri_cpe_ptmp Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Will my CPU/RAM be sufficient?
Hello! I'm new to asterisk and linux, so please don't blame me if I write silly things :-) I'd like to setup a system with IVR only. I'll use a SIP gateway to receive calls from the outside world, and I'll install asterisk on a dedicated linux server placed in another location that will be permanently connected to the gateway through the internet. I expect to have 8 concurrent calls maximum. Now, I have to choose the server that I will rent, and I don't want to take a machine which is too powerful for my needs in order not to waste money. Luckily many of you say that linux and asterisk are so performant that you normally do not need such a powerful CPU... One of the possibilities is: Celeron 1.7 GHz 128k cache HD 30/40GB U.ATA 100 128 MB DDR Ram network card 100Mbit/s M/B intel Then I can add memory (128 MB seems to be not too much to meI would put 256 MB). The gateway supports many codecs. If I use the gsm, do I save on CPU work? I mean: as asterisk uses gsm sound files, I wonder if using the gsm codecs the files are not treated on the fly but just read and sent directly to the gateway. What do you think about my thoughts? Will the computer be sufficient, or would you take a more powerful one? I can always take this computer and later take a more powerful one, but I would lose all the money of the setup, so I would like to take the good one since the beginning. Thank you, ciao Marco ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] zaphfc: empty HDLC frame or bad CRC received
Emanuele Pucciarelli wrote: Are you sharing the IRQ? (check /proc/interrupts) hi, what do you think? this is a bit too much low level for me. ciop:~# cat /proc/interrupts CPU0 0: 780877027IO-APIC-edge timer 7: 2IO-APIC-edge parport0 9: 1 IO-APIC-level acpi 12: 5968 IO-APIC-level VIA8233 14:1778879IO-APIC-edge ide0 15: 2IO-APIC-edge ide1 169: 66422031 IO-APIC-level eth0 177: 49271618 IO-APIC-level eth1 185: 1332694587 IO-APIC-level zaphfc NMI: 0 LOC: 780882796 ERR: 0 MIS: 0 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] zaphfc: empty HDLC frame or bad CRC received
Hi, i've downloaded/compiled/installed the bristuffed asterisk Asterisk 1.0.7-BRIstuffed-0.2.0-RC8a and i'm using it with an hfc card. It runs on a debian 3.1 sarge machine with kernel 2.6.11. Asterisk works well if i configure the card using isdn4linux. I'm having problems dialing out (not tried the input yet). This is the output from asterisk: -- Accepting AUTHENTICATED call from 192.168.0.5, requested format = 4, actual format = 4 -- Executing Dial(IAX2/[EMAIL PROTECTED]/1, Zap/g1/3***) in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called g1/3** -- Channel 0/1, span 1 got hangup May 25 11:11:41 WARNING[18260]: app_dial.c:412 wait_for_answer: Unable to forward voice -- Hungup 'Zap/1-1' == No one is available to answer at this time -- Executing Answer(IAX2/[EMAIL PROTECTED]/1, ) in new stack -- Executing Playtones(IAX2/[EMAIL PROTECTED]/1, congestion) in new stack -- Executing Congestion(IAX2/[EMAIL PROTECTED]/1, ) in new stack == Spawn extension (esterni, 3***, 4) exited non-zero on 'IAX2/[EMAIL PROTECTED]/1' -- Executing Goto(IAX2/[EMAIL PROTECTED]/1, default|h|1) in new stack -- Goto (default,h,1) -- Executing Hangup(IAX2/[EMAIL PROTECTED]/1, ) in new stack == Spawn extension (default, h, 1) exited non-zero on 'IAX2/[EMAIL PROTECTED]/1' -- Hungup 'IAX2/[EMAIL PROTECTED]/1' this is the message i get in the syslog (a lot of these messages): zaphfc: empty HDLC frame or bad CRC received (framelen = 5, stat = 0xff, card = 0). this is the zaptel.conf: loadzone=it defaultzone=it span=1,1,3,ccs,ami bchan=1-2 dchan=3 this is the zapata.conf: group = 1 context=default signalling = bri_net_ptmp channel = 1-2 this is the modules.conf: [modules] autoload=yes noload = pbx_gtkconsole.so noload = pbx_kdeconsole.so noload = app_intercom.so load = res_features.so load = chan_modem.so load = res_musiconhold.so load = chan_alsa.so noload = chan_oss.so load = chan_zap.so [global] chan_modem.so=yes chan_zap.so=yes this is the output from the module insertion: Zapata Telephony Interface Registered on major 196 PCI: Enabling device :00:06.0 ( - 0003) ACPI: PCI interrupt :00:06.0[A] - GSI 17 (level, low) - IRQ 185 zaphfc: CCD/Billion/Asuscom 2BD0 configured at mem 0xd08eaf00 fifo 0xcd248000(0xd248000) IRQ 185 HZ 1000 zaphfc: Card 0 configured for TE mode zaphfc: 1 hfc-pci card(s) in this box. this is the ztcfg -vvv: Zaptel Configuration == SPAN 1: CCS/ AMI Build-out: 399-533 feet (DSX-1) Channel map: Channel 01: Individual Clear channel (Default) (Slaves: 01) Channel 02: Individual Clear channel (Default) (Slaves: 02) Channel 03: D-channel (Default) (Slaves: 03) 3 channels configured. Thanks in advance ciao ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] VoiceXML
Hi, Anyone has a working example of VoiceXML with asterisk ? i was looking around voip-info and the internet, and couldnt find more then proof of concept documents. Also, does anyone knows how FWD does their VoiceXML (411) service ? Thanks for any info Marco. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Chan_modem_*
Hi, I was looking for solutions for simple FXO cards, and came across the two modem channels in the asterisk channels/ dir, i assume they are there becuase someone made these two types of modems work as FXO (or are they there for other purpose ?), does anyone have any info on these channels ? anyone has them working with any type of modem ? (aopen or bestdata). Marco. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Need info : lspci
Hi, I need some info from people with the x100p card (digium or clone), please send me the output of lspci and lspci -n from your linux machine, i am tring to find out something on my * server. Thanks. Marco. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
R: [Asterisk-Users] Voicemail SMS Alert - Possible?
Use externnotify (see http://www.voip-info.org/wiki-Asterisk+config+voicemail.conf) with a script to send sms. Some time ago I used a perl script called sendSms found in Internet. Bye. Marco -Messaggio originale- Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Per conto di Julius Kidubuka Inviato: lunedì 14 marzo 2005 09.09 A: asterisk-users@lists.digium.com Oggetto: [Asterisk-Users] Voicemail SMS Alert - Possible? I need to be able to send an sms alert to one's mobile/cell phone. For instance, when I receive a voicemail message in my inbox, I also want to be able to get a message on my cell phone alerting me of this e-mail. How possible is this? And if it is, what do I need to do to get the service up and running? Ideas are most welcome. Thanks, Julius. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] possible bug in chan_capi concerning context handling
Do you have an 's' extention in the default context ?
Marco.
Dimitris Kounalakis wrote:
Hello,
I am trying to configure asterisk 1.0.7pre to get incoming calls from an
ISDN line using an AVM fritz PCI 2.0 with Chan_capi 0.3.5. My problem is
that the context is not recognised in the /etc/asterisk/capi.conf
I have in /etc/asterisk/capi.conf 's section [interfaces] the
following directive
context=isdn
and the following directive in /etc/asterisk/extensions.conf in the
context [isdn]
[isdn]
exten = s,1,Dial(SIP/${DNID:4},60,tr)
Here follows the debug info I get when an incoming call starts:
== CONNECT_IND
(PLCI=0x101,DID=2810111694,CID=2810111694,CIP=0x1,CONTROLLER=0x1)
-- creating pipe for PLCI=0x101 msn = 2810111694
sent ALERT_REQ PLCI = 0x101
== Starting CAPI[contr1/2810111694]/3 at ,2810111694,1 failed so
falling back to exten 's'
== Starting CAPI[contr1/2810111694]/3 at ,s,1 still failed so falling
back to context 'default'
Mar 13 11:52:41 WARNING[10744]: pbx.c:1893 ast_pbx_run: Channel
'CAPI[contr1/2810111694]/3' sent into invalid extension 's' in context
'default', but no invalid handler
-- CAPI Hangingup
-
When I move the exten = s,1,Dial(${DNID:4},60,tr) in the context
[default] of the /etc/asterisk/extensions.conf, I get the following
debug info and the sip phone rings ok:
--
== CONNECT_IND
(PLCI=0x101,DID=2810111694,CID=2810111694,CIP=0x1,CONTROLLER=0x1)
-- creating pipe for PLCI=0x101 msn = 2810111694
sent ALERT_REQ PLCI = 0x101
== Starting CAPI[contr1/2810111694]/4 at ,2810111694,1 failed so
falling back to exten 's'
== Starting CAPI[contr1/2810111694]/4 at ,s,1 still failed so falling
back to context 'default'
-- Executing Dial(CAPI[contr1/2810111694]/4, SIP/111694|60|tr) in
new stack
-- Called 111694
--
Is this a bug? It does not handle the context, so, it can not find what
to do, it works only with the default context.
Thank you in advance,
Dimitris
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] zaphfc error
In data Tue, 8 Mar 2005 18:25:38 + (GMT), hai scritto: [chan_zap.so]Mar 8 17:53:06 WARNING[2447]: loader.c:258 ast_load_resource: /usr/lib/asterisk/modules/chan_zap.so: undefined symbol: ast_retrieve_call_to_death Mar 8 17:53:06 WARNING[2447]: loader.c:391 load_modules: Loading module chan_zap.so failed! I'm using asterisk from debian/sid: Asterisk 1.0.5-BRIstuffed-0.2.0-RC7e zaptel modules are version 1.0.4 zaphfc is from bristuff-0.2.0-RC7e and it's compiled against zaptel source version 1.0.4 - - - - - - 8 snipped Change your: load = chan_zap.so load = res_musiconhold.so ok, this worked. I had to load also chan_modem and i had to fix a missing [channels] in zapata.conf now i get: -- Executing Dial(IAX2/[EMAIL PROTECTED]/2, Zap/g1/the number) in new stack -- Called g1/the number -- Channel 0/1, span 1 got hangup Mar 9 15:09:53 WARNING[5329]: app_dial.c:415 wait_for_answer: Unable to forward voice -- Hungup 'Zap/1-1' My asterisk setup works well with isdn4linux so i think that the problem relies in the zaphfc setup. ciao ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] sip hangup detection problem
Hi ml, I'm experiencing some problem detecting hangup with sip channel. I have an asterisk on remote site behind NAT and two xlite at home behind nat. I can make calls between them but hangup cannot be detected. When I try to hangup a call I see xlite that tell me hanging up for some seconds and hangups the call but the other side still be connected.. I also see on asterisk cli this message: chan_sip.c:787 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 45854 (Non-critical Response) Does someone experience the same problem? Can someone help me? Thanks. Marco Ziglioli ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] zaphfc error
I have some problems starting asterisk with a hfc card using zaphfc: [chan_zap.so]Mar 8 17:53:06 WARNING[2447]: loader.c:258 ast_load_resource: /usr/lib/asterisk/modules/chan_zap.so: undefined symbol: ast_retrieve_call_to_death Mar 8 17:53:06 WARNING[2447]: loader.c:391 load_modules: Loading module chan_zap.so failed! I'm using asterisk from debian/sid: Asterisk 1.0.5-BRIstuffed-0.2.0-RC7e zaptel modules are version 1.0.4 zaphfc is from bristuff-0.2.0-RC7e and it's compiled against zaptel source version 1.0.4 I do things in this order: modprobe zapata modprobe zaphfc ztcfg -vv asterisk -c rmmod zaphfc rmmod zapata Here are the config files: ### zaptel.conf: span=1,1,3,ccs,ami bchan=1-2 dchan=3 # zapata.conf: switchtype = euroisdn signalling = bri_net_ptmp pridialplan=local echocancel=yes immediate=yes group = 1 context=local channel = 1-2 ## modules.conf [modules] autoload=yes noload = pbx_gtkconsole.so ;load = pbx_gtkconsole.so noload = pbx_kdeconsole.so noload = app_intercom.so noload = chan_modem.so load = chan_zap.so load = res_musiconhold.so noload = chan_alsa.so noload = chan_oss.so [global] chan_modem.so=no chan_zap.so=yes ## KERNEL MODULE INSERTION MESSAGES AND ZTCFG: ciop:~# modprobe zaptel;modprobe zaphfc;tail -20 /var/log/kern.log;ztcfg -vv Mar 8 17:56:06 ciop kernel: Zapata Telephony Interface Registered on major 196 Mar 8 17:56:06 ciop kernel: PCI: Enabling device :00:06.0 ( - 0003) Mar 8 17:56:06 ciop kernel: ACPI: PCI interrupt :00:06.0[A] - GSI 17 (level, low) - IRQ 169 Mar 8 17:56:06 ciop kernel: zaphfc: CCD/Billion/Asuscom 2BD0 configured at mem 0xd081ff00 fifo 0xc6628000(0x6628000) IRQ 169 HZ 1000 Mar 8 17:56:06 ciop kernel: zaphfc: Card 0 configured for TE mode Mar 8 17:56:06 ciop kernel: zaphfc: 1 hfc-pci card(s) in this box. Mar 8 17:56:27 ciop kernel: zaphfc: stop Mar 8 17:56:27 ciop kernel: zaphfc: shutting down card at d081ff00. Mar 8 17:56:27 ciop kernel: unregistered from zaptel. Mar 8 17:56:27 ciop kernel: zaphfc: freed one card. Mar 8 17:56:27 ciop kernel: Zapata Telephony Interface Unloaded Mar 8 17:56:58 ciop kernel: Zapata Telephony Interface Registered on major 196 Mar 8 17:56:58 ciop kernel: PCI: Enabling device :00:06.0 ( - 0003) Mar 8 17:56:58 ciop kernel: ACPI: PCI interrupt :00:06.0[A] - GSI 17 (level, low) - IRQ 169 Mar 8 17:56:58 ciop kernel: zaphfc: CCD/Billion/Asuscom 2BD0 configured at mem 0xd081ff00 fifo 0xc66c8000(0x66c8000) IRQ 169 HZ 1000 Mar 8 17:56:58 ciop kernel: zaphfc: Card 0 configured for TE mode Mar 8 17:56:58 ciop kernel: zaphfc: 1 hfc-pci card(s) in this box. Zaptel Configuration == SPAN 1: CCS/ AMI Build-out: 399-533 feet (DSX-1) Channel map: Channel 01: Individual Clear channel (Default) (Slaves: 01) Channel 02: Individual Clear channel (Default) (Slaves: 02) Channel 03: D-channel (Default) (Slaves: 03) 3 channels configured. ## KERNEL MODULE REMOVAL ciop:~# rmmod zaphfc;rmmod zaptel;tail -20 /var/log/kern.log Mar 8 18:02:09 ciop kernel: zaphfc: stop Mar 8 18:02:09 ciop kernel: zaphfc: shutting down card at d081ff00. Mar 8 18:02:09 ciop kernel: unregistered from zaptel. Mar 8 18:02:09 ciop kernel: zaphfc: freed one card. Mar 8 18:02:09 ciop kernel: Zapata Telephony Interface Unloaded ## ERRORS FROM THE KERNEL MODULE Mar 8 17:57:46 ciop kernel: zaphfc: bchan rx fifo not enough bytes to receive! (z1=528, z2=527, wanted 8 got 2), probably a buffer overrun. Mar 8 17:58:34 ciop kernel: zaphfc: bchan rx fifo not enough bytes to receive! (z1=528, z2=527, wanted 8 got 2), probably a buffer overrun. Mar 8 17:59:23 ciop kernel: zaphfc: bchan rx fifo not enough bytes to receive! (z1=528, z2=527, wanted 8 got 2), probably a buffer overrun. Mar 8 18:00:11 ciop kernel: zaphfc: bchan rx fifo not enough bytes to receive! (z1=528, z2=527, wanted 8 got 2), probably a buffer overrun. Mar 8 18:01:00 ciop kernel: zaphfc: bchan rx fifo not enough bytes to receive! (z1=528, z2=527, wanted 8 got 2), probably a buffer overrun. Mar 8 18:01:48 ciop kernel: zaphfc: bchan rx fifo not enough bytes to receive! (z1=528, z2=527, wanted 8 got 2), probably a buffer overrun. TIA ciao ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk supports VXML?
Hi all where can I find infos aboutthis VXML intepreterfor asterisk? Thanks Marco Hi Foong, That's a good question you've put out there. Yes, Asterisk supports VXML andhere's how it's done; Firstly in the SIP.conf, you need to have your VXML application/browserdefined; sip.conf: [vxmlapp] type=friend insecure=yes username=777 reinvite=no host=123.45.67.8 Then in the EXTENSIONS.conf it will look like this; extensions.conf: exten =777,1,Setvar,VXML_URL=voicexml=http%3A%2F%2F123.45.67.20%3A6969%2Fhellovxml%2Fhellovxml exten = 777,2,Dial,sip/vxmlapp|10 exten = 777,3,HangUp Hope this'll clear your thoughts. Cheers! Lilantha Karunaratne MSCSTel: (65) 90403497 _ From: asterisk-users-bounces at lists.digium.com[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Chee FoongSent: Friday, February 25, 2005 10:17 AMTo: asterisk-users at lists.digium.comSubject: [Asterisk-Users] asterisk supports VXML? Hello,Does asterisk supports VXML?Couldn't find much resource on that on google and wiki.ThanksFoong ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] signaling problems
Hi ml, this is my problem: I have an Asterisk on remote site (my office) and two x-lite at home behind a ful cone nat. Both my ua can register, I can place and receive calls from both the phones and I can hear voice, so I don't think I have nat problem but when when i place a call if the called party hangup, calling party doesn't receive the signal and it stays connected. I also experienced the same problem placing a call on hold. Calling party can place the call on hold (called party listen moh) but called party cannot do it. Watching asterisk CLI no called party signals were detected? Why these? Can someone help me? Regards. Marco Ziglioli Alascom Services S.R.L. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP registration problem
Hi, I am adding phones to my asterisk setup, until now i worked with some softphones, with no problem, I got some Grandstream BT100 phones, and see something strange in the log, the on the phone's screen, This is from the log : Found peer '122' Looking for 122 in default Transmitting (no NAT): SIP/2.0 404 Not Found This happends when the action is SUBSCRIBE , Now, this is a SIP client, defined in the sip.conf, as [122] context=default ... and also the exten is in the default context in the extension conf file, Right after the the peer seems to be registered, and the phone seems to work, but from time to time, i see 404 on the phone's display, and need to touch it to make it change (dial something, or just pick up and hangup) I couldnt find why this is happening, i searched, and found some with the same problem, but no solution, If you have any idea why this is happening, i will be glad to hear it. Thanks. Marco. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk and #
Hi ml, I have a problem related to call parking. When on my X-Lite try to parking a call dialing #700 I don't obtain anything. I can only ear dtmf tones during conversation but not other happens. I also read in some post that only pressing # should place call in hold state but this doesn't happen on my system. Can someone help me? Thanks. Marco ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
R: [Asterisk-Users] Asterisk and #
Title: Messaggio Ok! problem solved! tT missed on extension used for test. Thank you very much for support Marco -Messaggio originale-Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Per conto di Dennis WebbInviato: giovedì 24 febbraio 2005 19.05A: Asterisk Users Mailing List - Non-Commercial DiscussionOggetto: Re: [Asterisk-Users] Asterisk and #make sure you have tT when the incoming call comes in.I've been studying up on parking today and saw this a few times.On Thu, 2005-02-24 at 11:51, Marco Ziglioli wrote: Hi ml, I have a problem related to call parking. When on my X-Lite try to parking a call dialing #700 I don't obtain anything. I can only ear dtmf tones during conversation but not other happens. I also read in some post that only pressing # should place call in hold state but this doesn't happen on my system. Can someone help me? Thanks. Marco ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAXTel problems
Hi, I tried to add the IAXTel config to my asterisk, so i can dial free numbers inside the US from my SIP softphone (X-lite), everything seems to be working, but the sound quality is terrible, the other side sounds like a digitized voice, and the voice is cut, i cant hear a full word, I tried using FWD IAX interface, and no problem there, it works great. Now, although this is in a testing phase, i wanted to know if i am missing something, or IAXTel is just problematic . I am dialing from Israel, over a E1 line, dont know exactly how much of my E1 reaches the US, but should be sufficent for one session (for which FWD works fine with) Any help appriciated. Marco. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Minimal hardware requirements
I have two comments: a. It maybe doesn't work because of the PCI specifications the box support. If was manufactured before Jan 2000, it is quite probably that it won't recognize the Digium cards. b. From the point of view of load, I see no problems, I think the specs of the machine are enough for such a small system. Hope it helps. Marco -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Rudolf Ladyzhenskii Sent: Monday, February 21, 2005 6:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Minimal hardware requirements Hi, all I am doing prrof of concept system. I will have two IP phones connected to Asterisk box. Box itself will have 1 PSTN conenction and one analog phone conenction. A basic minimal configuration. At the moment I am planning to use an old PII-350 with 128M of RAM I have lying around. I can not test anything yet, as I am waiting for phones to arrive, so question is will that be enough to demonstrate? Thanks, Rudolf ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Zaptel Needed
You don't need the zaptel library if you aren't going to use any digium cards. Marco -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of [EMAIL PROTECTED] Sent: Thursday, February 17, 2005 8:02 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Zaptel Needed Hello All, Can someone please tell me about Zaptel? Is it only needed if you are going to have an interface card like TDM400P installed on the Asterisk server? Do you really need it if you do not have the interface card? Thanks, Lonnie ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] E1 and/or Euro-ISDN specifications?
Go to the ITU website www.itu.org there you can buy all the specifications you're looking for. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Daniel Nyström Sent: Tuesday, February 15, 2005 8:47 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] E1 and/or Euro-ISDN specifications? Where can I get E1 and/or Euro-ISDN specifications/data sheets? Are there specs for other E./G./Q./etc. protocols as well? Thanks! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] A hypothetical question...
The complete configuration of such a system requires a lot more of information that the one you gave.But, at a glance, Asterisk + SER is a good choice for this kind of venture. Asterisk can serve as the PSTN gateway (ISDN PRI connections primarily) and Voicemail server. SER can manage the billing and the VOIP-client part. You can mount as many as Asterisk and SER servers as much as your traffic will require. So, you don't have to spend a lot of $$$ to mount such a large implementation. As I mention earlier, this is just a fast glimpse to a complete solution, I personally have such an implementation, and let me tell you that this works really great!!! Hope this helps Marco -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of Rod BaconSent: Tuesday, February 15, 2005 4:30 PMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] A hypothetical question... I know this is casting a wide net, but If you were charged with building a large, public VOIP network with multiple PSTN gateways, the capacity to carry a lot of traffic and bill clients accurately, what pieces (brands, makes, models) would you use to assemble the solution? Assume that $$$ is not an issue. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] newbie: help two cisco phones (sip)
Have you set your DNS SRV entry for SIP correctly??? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Andrew White Sent: Tuesday, February 15, 2005 7:04 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] newbie: help two cisco phones (sip) Hi, I have two cisco phones with sip images and I am trying to configure to work with asterisk. Both can call demo numbers and voicemail etc. but can't call each other. sip show registry and sip show users both indicate that asterisk doesn't know the phones ip addresses, and when u try to place a call, it forwards to unanswered voicemail immediately. I have tried user_info: ip and also phone, but can't seem to get the phones to register. sip.conf has host=dynamic for both phones SIP image is version 7 anyone able to tell me where i'm going wrong ? tks Andrew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] connect asterisk to ISDN in China
Dear Xu, my name is Marco Castillo, I'm in Guatemala, Central America, and I have recently succesfully installed a TE110P here in Guatemala. There are many implementations of a E1 or T1, but I think that the great majority can be configured via the zaptel drivers. I will suggest you to buy a card and make the leap of faith!!! Regards Marco -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Xu, Duo Sent: Sunday, February 13, 2005 12:58 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] connect asterisk to ISDN in China Hi, I plan to install asterisk and connect it to telco through ISDN in China. I'd love to know if the ISDN standard in China has any difference than in America before I buy the digium card. anybody has experience in it? or anybody who installed asterisk with ISDN in asia can share their expierience? Or, can anybody give me some links to educate me ISDN knowledge about the difference in China? (My heard there is something different there, but i dont know the details.) Thanks __ Do you Yahoo!? The all-new My Yahoo! - What will yours do? http://my.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_capi and asterisk
I don't know about your problem, but since you use mISDN, why not use the specific chan_mISDN? http://www.beronet.com/?PageID=3017 It's Free Software (GPL) Regards Marco Menardi btw, if you login in their bug tracker, the home page has alink to a document that tells you how install their boards, mISDN and, AFAIR, use their chan_mISDN with asterisk. Anabela Abreu wrote: Hello, list a have a problem i can start asterisk, i get the fowlling error: [chan_capi.so] = (Common ISDN API for Asterisk) == Parsing '/etc/asterisk/capi.conf': Found Feb 11 13:50:36 NOTICE[2535]: chan_capi.c:2636 load_module: CAPI not installed! Feb 11 13:50:36 WARNING[2535]: loader.c:345 ast_load_resource: chan_capi.so: load_module failed, returning -1 Feb 11 13:50:36 WARNING[2535]: chan_capi.c:2812 unload_module: Unable to unregister from CAPI! == Unregistered channel type 'CAPI' Feb 11 13:50:36 WARNING[2535]: loader.c:391 load_modules: Loading module chan_capi.so failed! my lsmod shows: Module Size Used by mISDN_capi 85312 0 kernelcapi 45088 1 mISDN_capi hfcpci 28716 0 mISDN_dsp 197248 0 l3udss132008 0 mISDN_l2 38272 0 mISDN_l1 10632 0 mISDN_core 77732 6 mISDN_capi,hfcpci,mISDN_dsp,l3udss1,mISDN_l2,mISDN_l1 md5 4352 1 ipv6 235840 24 parport_pc 25024 1 lp 12396 0 parport42696 2 parport_pc,lp dm_mod 55444 0 uhci_hcd 31896 0 3c59x 36776 0 floppy 59568 0 ext3 116744 2 jbd74904 1 ext3 and my modules.conf : load = chan_capi.so [global] chan_capi.so=yes what seems to be the problem can someone help me? tahnk´s ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk not accepting multiple SIP phone logins
Remember that SIP uses DNS SRV entries, maybe one of the phones you use efectively use the DNS SRV entry and the other not. Some VoIP phones have a flag where you can deactivate this functionality for SIP. If not, make sure you have in your local DNS a SRV entry for SIP. Hope this helps. Marco -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Juki Sent: Thursday, February 10, 2005 11:08 PM To: [EMAIL PROTECTED] Cc: asterisk-users@lists.digium.com; asterisk-dev@lists.digium.com Subject: [Asterisk-Users] Asterisk not accepting multiple SIP phone logins Hi all, I have Asterisk running on FreeBSD 4.x and I have made configurations to sip.conf, extensions.conf and voicemail.conf. I have also setup SIP phones on two different PCs. My problem is that when one of the SIP phones logins in, the other won't. My sip.conf has: [101] type=friend host=dynamic username=101 secret=test dtmfmode=rfc2833 context=from-sip mailbox=201 callerid=101 2125 nat=yes My extensions.conf has: exten = 101,1,Dial(SIP/101,20,tr) exten = 101,2,VoiceMail,u101 exten = 101,102,VoiceMail,b101 My voicemail.conf has: 101 = 2348,Emma, [EMAIL PROTECTED] Any ideas are most welcome. -- Rgds, Juki ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] No dialtone in a E1
Thank you Peter, how can I add the options to Dial to generate ringback??? do you have an example??? By the way, it is a PRI E1, with 30 bchannels and 1 dchannel. For a little background, I'm intending to replace my actual PBX with Asterisk, and everything is just working fine, until yesterday when I realized that when a call was made from some external lines, this lines didn't receive a dialtone. For this reason, I began to make some exhaustive test cases, and began to make calls from distinct providers to my E1. In all this testing I received a dialtone, except for a GSM cellular phone from a specific Telco. I tested some others GSM cellulars from the same Telco, and got always the same functionality, they didn't receive a dialtone. I think that if Asterisk can generate a ringback, this is going to solve all my problems with this little issue. Thank you in advance Peter for your help. Marco -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Peter Svensson Sent: Thursday, February 10, 2005 6:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] No dialtone in a E1 On Thu, 10 Feb 2005, Marco Castillo wrote: Hi, I'm having a little problem when trying to make a call from asterisk. I connect a SIP phone to asterisk, and in the asterisk box I have a TE110P card connected to a E1. When a SIP client makes a call through the E1, I received no dialtone in the SIP client. In the same manner, when somebody from the POTS network makes a call to a SIP client (through * and the E1) he doesn't receive the apropiate tone of call progress. Does anyone has some ideas about this? Are you talking about an ISDN E1 or another form of E1? On isdn dialtone is an optional feature of the specification and there are many implementations of isdn. I think it is mandatory on EuroISDN. Since asterisk normally generates the dialtone itself there should be little nead for the dialtone from the pstn. We use the dialtone from the network ourselves, but asterisk could provide it as well. In band call progress is also a feature of the net on isdn. If the net does not provide it you will have to do so yourself. Just add the proper options to Dial to generate ringback and if the call fails you generate the matching sound (Busy etc). Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Video Conference
Hi Florian, thanks for your help. Yes I have enable videosuport in the sip.conf, and I think that i have the proper codecs. This is what i have in my sip.conf... [general] context=default videosupport=yes [097] type=friend username=video secret=video host=dynamic callerid=Video 097 canreinvite=no disallow=all ;allow=ulaw ;allow=alaw ;allow=speex allow=gsm allow=h261 allow=h263 nat=yes context=ip ;qualify=yes ;dtmfmode=rfc2833 Thanks for any help Marco González __ Do you Yahoo!? The all-new My Yahoo! - What will yours do? http://my.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Debian way of compiling zaptel kernel modules
Be sure you're using Debian with kernel 2.4.20 or more (if you use the stable release version of Debian, using dselect will just upgrade your kernel until 2.4.18, and the zaptel libraries won't be properly compiled). Using dselect o apt, get the kernel-sources packages and the kernel-heades packages. And modify the Makefile for the zaptel libraries to point to the proper directories for the kernel headers (don't use usr/include/linux). Hope this helps Marco -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Geoff Nordli Sent: Thursday, February 10, 2005 11:56 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Debian way of compiling zaptel kernel modules I ran apt-get -b source zaptel-source to download and compile the zaptel source. It successfully compiles and builds the following packages: libtonezone-dev_1.0.2-2_i386.deb libtonezone1_1.0.2-2_i386.deb zaptel-source_1.0.2-2_all.deb zaptel_1.0.2-2_i386.deb None of them contain the kernel modules. Is there a way I can get it to compile the kernel modules? Thanks, Geoff ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] No dialtone in a E1
Hi, I'm having a little problem when trying to make a call from asterisk. I connect a SIP phone to asterisk, and in the asterisk box I have a TE110P card connected to a E1. When a SIP client makes a call through the E1, I received no dialtone in the SIP client. In the same manner, when somebody from the POTS network makes a call to a SIP client (through * and the E1) he doesn't receive the apropiate tone of call progress. Does anyone has some ideas about this? Ing. Marco Antonio Castillo Chief Design Engineer Van Der Kaaden IT Consulting Guatemala, Guatemala C.A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk Compile Problem on Red Hat 9 resolved
For softphones, I used SJPhone, is a very good SIP phone, and I have it working on asterisk. The setup is kind of tricky, 'cause you must remember to set the sip register in your local DNS. For a good overview and introductory tutorial to asterisk, go to the asterisk home site (www.asterisk.org), go to resources, there you will find good introductory material. Marco -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of vdasilvaSent: Wednesday, February 09, 2005 1:02 PMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] Asterisk Compile Problem on Red Hat 9 resolved Thanks Noah I got the source with CVS to a Windows machine, this is the source causing the problem, although I suspect that getting the files to Windows and then copying them to Linux was not a good idea. I then got the tarball files, unzipped them on Linux and compiled and everything installed fine. My next goal is to setup 1 SIP channel, and be able to call the Asterisk PBX from a softphone. Then setup 2 SIP channes and be able to call one from another. What is the best open source softphone software available for this? And what is the best documentation source for finding out how to setup the channesl and Asterisk in general? Vince I get the following error when trying to compile asterisk 1.05 on red hat 9. [EMAIL PROTECTED] asterisk]# make install *** You don't have mpg123 installed. You're going to need *** *** it if you want MusicOnHold *** ./mkdep -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -02/08/05-20:18:18\" -DINSTALL_PREFIX=\"\" -DASTETCDIR=\"/etc/asterisk\" -DASTLIBDIR=\"/usr/lib/asterisk\" -DASTVARLIBDIR=\"/var/lib/asterisk\" -DASTVARRUNDIR=\"/var/run\" -DASTSPOOLDIR=\"/var/spool/asterisk\" -DASTLOGDIR=\"/var/log/asterisk\" -DASTCONFPATH=\"/etc/asterisk/asterisk.conf\" -DASTMODDIR=\"/usr/lib/asterisk/modules\" -DASTAGIDIR=\"/var/lib/asterisk/agi-bin\" -DBUSYDETECT_MARTIN `ls *.c` : invalid option Usage: /bin/sh [GNU long option] [option] ... /bin/sh [GNU long option] [option] script-file ... GNU long options: --debug --dump-po-strings --dump-strings --help --init-file --login --noediting --noprofile --norc --posix --rcfile --rpm-requires --restricted --verbose --version --wordexp Shell options: -irsD or -c command or -O shopt_option (invocation only) -abefhkmnptuvxBCHP or -o option make: *** [.depend] Error 2 Any help is greatly appreciated Vince ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Video Conference
Hi everyone!!! My Name is Marco, I'm from Caracas,Venezuela. I'm a new Asterisk user... I'm trying to make a video conference with sip. I have the Eyebeam from Xten, a video sip phone. I have a good audio conection, but nothing about the video Now I'm trying to do the same with h323, but don't know how to compile and configure the modules. Can anybody give me a help with this??? Thanks for everything... Marco __ Do you Yahoo!? Meet the all-new My Yahoo! - Try it today! http://my.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] No dial tone...
I recently have purchased a new TE110P card, that provides a single T1/E1 port. I have installed it and everything works fine, except for the dial tones. When I made a call from a SIP phone to a channel in the TE110P, I receive no dial tone. When I receive a call in a SIP phone from a channel in the TE110P, I have no dial tone in the caller phone. Does anybody has a idea???, Is this configurable in the zaptel.conf file??? Any help would be greatly apreciatted. Ing. Marco Antonio Castillo Chief Design Engineer Van Der Kaaden IT Consulting Guatemala, Guatemala C.A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
