Re: [asterisk-users] Microsoft launches first PABX
Christopher Chan wrote: C F wrote: I think yes, why you disagree? Has Microsoft actually ever come with such useful features? It would be great to demonstrate the complete instability/insecurity of Windows based servers by have it shut down automatically in front the boss with a recorded message :D. Even better if it comes with a BSOD command :) It's doable with Vista, but you need voice recognition turned on first: http://blogs.zdnet.com/Ou/?p=416 -- James Andrewartha ___ --Bandwidth 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] Teliax problems, they say use SIP, more mature better working than IAX
On 19 Mar 2007, at 20:51, Scott Plante wrote: Raw Hangup The code says: /* A call arrived for a nonexistent destination. Unless it's an inval frame, reply with an inval */ If you can, produce a packet dump of a failing call with ethereal - or with iax2 debg, send it to me with your iax.conf and I'll take a look. Tim Panton www.mexuar.net www.westhawk.co.uk/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dell poweredge 860 acceptable for officeenvironment ?
On 20 Mar 2007, at 03:14, Leif Neland wrote: Steve Totaro wrote: Stephen Bosch wrote: Olivier wrote: I'm really after 1U-2U silent servers as I've got the feeling most of them are too noisy for offices and most of our clients don't have server rooms. Try this: http://www.tomshardware.com/2006/01/09/strip_out_the_fans/ -s The fans are in there for a reason. It appears you haven't read the article. The tomshardware-guys (no gals would do this...) have removed the fans, and immersed the innards of the computer in a sealed cabinet filled with cooking oil. So they have a completely silent machine in 40C warm oil. Amazing... I had a friend who filled her calculator with warm molasses, but that was an accident :-) Tim Panton www.mexuar.net www.westhawk.co.uk/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] how to interconnection asterisk(sip) with mera
dear all,
we need help for integration asterisk (sip) with mera
we have configure for sip.conf and extentions.conf
sip.conf
[mvts]
context=mvts
type=friend
host=10.10.0.2
dtmf=rfc2833
in extentions.conf
[mvts]
;
; mvts
exten = _01162.,1,SetCallerID(mvts)
exten = _01162.,2,SetCIDName(to mvts)
exten = _01162.,3,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
i need if i dial 01162 in mera replace with 44522(interlocal call)
but when i dial 01162(interlocal call) in asterisk not respond.
many thx for your help
best regards
yono
http://dhidhit.blogspot.com/
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[asterisk-users] Which parameters of a live Asterisk server would you monitor ?
Hi, Let's say you have an Asterisk server running. Which parameters would you check to improve service continuity ? I was thinking of : - telco lines status (make sure every is up) - registered hardphones - config files backup (compare live and saved configuration files, if files differ, notifies the administration team) - systems variables (disk and CPU) - log files (trigger an alarm for every ERROR or NOTICE message in full logs) What do you think ? Regards ___ --Bandwidth 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] Configuring Faxs any help :)
and this is the /var/spool/hylafax/log/c1: http://pastebin.ca/403282 cat /var/spool/hylafax/log/c3 :: http://pastebin.ca/403291 thank you ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Configuring Faxs any help :)
that's work know i don't know where is the problem but i folowed the links bellow: http://www.guardiani.us/index.php/TrixBox_IAXModem_HylaFax#Satisfy_HylaFax_Deps http://www.ecualug.org/?q=2007/02/27/comos/como_recibir_fax_en_asterisk http://threebit.net/mail-archive/asterisk-users/msg04703.html thank you; ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Zapateller not playing audio via SIP Trunk?
Hi All I'm tracing a very strange problem which I could reproduce with different versions up to 1.2.5 (sorry, didn't update to a newer one yet). Scenario 1: Problem does not occure. = Sip Phone registered directly to the Asterisk. exten = i,1,Zapateller() exten = i,n,Playback(invalid,noanswer) exten = i,n,Hangup() Works like expected. I dial an invalid number, hear the SIT tone and then the Announcement. Scenario 2: Problem does occure. === SIP 'trunk' to another SIP PBX (or another Asterisk). exten = i,1,Zapateller() exten = i,n,Playback(invalid,noanswer) exten = i,n,Hangup() Does not work, Zapateller seams to be 'hanging' forever without playing any audio. exten = i,1,Zapateller(answer) exten = i,n,Playback(invalid,noanswer) exten = i,n,Hangup() Well, this does work, but the customer would have to pay as a CDR is generated. We want Early Audio. exten = i,1,Playback(invalid,noanswer) exten = i,n,Hangup() Strangely, this again works, so it is not a 183 Proccessing (Early Audio) problem, but seams to be a Zapateller Application Problem... exten = i,1,Playback(invalid,noanswer) exten = i,n,Zapateller() exten = i,n,Hangup() Early Audio Announcement is played, but then again as soon as Zapateller is executed, it 'hangs'. Any idea what causes Zapateller to hang if it should play early audio via a SIP Trunk? Mit freundlichen Grüssen Benoit Panizzon -- I m p r o W a r e A G-System Services __ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 PrattelnFax +41 61 826 93 01 Schweiz Web http://www.imp.ch __ pgpGEbnans9YI.pgp Description: PGP signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] transfer=mediaonly : can't hear nothing
Kevin P. Fleming ha scritto:
OK, then you'll need to get a verbose/debug console trace, and
preferably a packet capture of the IAX2 traffic on 'Server', and post a
bug on bugs.digium.com with those files attached.
___
While setting up the servers to gather the logs I've tryed a
configuration which is so hello world it seems unprobable to me it
can't work due to a bug.
I post once again here, sorry for the verbosity, if then in your opinion
there's still something wrong with * internals and not with my
understanding of the configs I'll open the bug.
I anticipate that only with mediaonly (when I can't hear) I get these
messages : Received iseqno 4 not within window 5-5 which seems to
remand to bug number 0006808, but I've tested also with jitterbuffer=no
on all machines and the problem remains.
Also I get some Subclass: (38?) packets, only in mediaonly mode.
3 machines, all on the same class C net (192.168.52.x), 2 are clients
(C001 and C002) and one is the server
C001 has two nics, the second being 192.168.0.1 connected to a switch
with a linksys pap in it, which generates the call:
C001 and C002 sip.conf, iax.conf and extensions.conf are the same
(except of course for IPs where to listen and credentials)
C00x sip.conf:
[general]
context=default ; Default context for incoming calls
realm=retireti.it
bindport=5060 ; UDP Port to bind to (SIP standard port
is 5060)
bindaddr=192.168.0.1; IP address to bind to (0.0.0.0
binds to all)
srvlookup=no
tos_sip=cs3; Sets TOS for SIP packets.
tos_audio=ef ; Sets TOS for RTP audio packets.
disallow=all
allow = alaw
language=it
dtmfmode = inband
progressinband=no
canreinvite=no
qualify=yes
jbenable = no
jbforce = no
jbmaxsize = 400
jbimpl = adaptive
[0100x01]
type=friend
secret=0100x00
context=outgoing
callerid=(whatever 0100x01)
host=dynamic
C00x iax.conf:
[general]
bindport=4569
bindaddr=192.168.52.9x (C001 .94 and C002 .95)
language=it
disallow=all
allow = alaw
allow = gsm
jitterbuffer = yes
forcejitterbuffer = no
maxjitterbuffer = 400
dropcount=2
maxjitterinterps=10
resyncthreshold=1000
maxexcessbuffer=80
minexcessbuffer=10
jittershrinkrate=1
autokill=yes
auth=md5
register = 0100x01:[EMAIL PROTECTED]
[server]
type=friend
context=incoming
secret=pwd
auth=md5
host=192.168.52.56
disallow=all
allow=alaw
allow=gsm
C00x extensions.conf :
[general]
static = yes
writeprotect = no
clearglobalvars = no
[globals]
CODACCOUNT = 0100x01
PWD = 0100x00
SERVER = 192.168.52.56
[outgoing]
exten = _X.,1,NoOp(esco)
;exten = _X.,n,Dial(IAX2/${CODACCOUNT}:[EMAIL PROTECTED]/${EXTEN})
exten = _X.,n,Dial(IAX2/${CODACCOUNT}:[EMAIL PROTECTED]/${EXTEN})
exten = _X.,n,Hangup
[incoming]
exten = _X.,1,NoOp(entro)
exten = _X.,n,Answer
exten = _X.,n,Playback(tt-weasels)
exten = _X.,n,Echo
exten = _X.,n,Hangup
now Server configs :
iax.conf :
[general]
bindport=4569
bindaddr=192.168.52.56
language=it
disallow=all
allow=alaw
allow=gsm
jitterbuffer = yes
forcejitterbuffer = no
maxjitterbuffer = 100
dropcount=2
maxjitterinterps=10
resyncthreshold=1000
maxexcessbuffer=80
minexcessbuffer=10
jittershrinkrate=1
context=default
autokill=yes
[0100101]
username=0100101
type=friend
secret=0100100
auth=md5
host=dynamic
context=default
callerid=0100101
transfer=no
qualify=yes
[0100201]
username=0100201
type=friend
secret=0100200
auth=md5
host=dynamic
context=default
callerid=0100201
transfer=no
qualify=yes
extensions.conf
[general]
static=yes
writeprotect=no
clearglobalvars=no
[globals]
[default]
exten = _X.,1,NoOp(here we are)
exten = _X.,n,Dial(IAX2/server:[EMAIL PROTECTED]/${EXTEN})
exten = _X.,n,Hangup
As you can see I've removed the realtime engine, and I've no input
client and termination clients difference, C001 calls the server,
which calls C002, which playback something and then Echoes, anyway both
C001 and C002 are the same type of registered, monitored friends for
the Server.
transfer=no, and all works ok, with debug,verbose and 'iax2 set debug' I
see in Server's CLI :
*CLI Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX
Subclass: NEW
Timestamp: 00010ms SCall: 6 DCall: 0 [192.168.52.94:4569]
VERSION : 2
CALLED NUMBER : 12
CODEC_PREFS : (alaw|gsm)
CALLING NUMBER : 0100101
CALLING PRESNTN : 0
CALLING TYPEOFN : 0
CALLING TRANSIT : 0
CALLING NAME: whatever
LANGUAGE: it
USERNAME: 0100101
FORMAT : 8
CAPABILITY : 57354
ADSICPE : 2
DATE TIME : 2007-03-20 12:16:30
Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass:
AUTHREQ
Timestamp: 00013ms SCall: 3 DCall: 6 [192.168.52.94:4569]
AUTHMETHODS : 2
CHALLENGE : 347981677
USERNAME: 0100101
Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass:
AUTHREP
Timestamp: 00030ms SCall: 6
RE: [asterisk-users] Which parameters of a live Asterisk server wouldyou monitor
From: Olivier [EMAIL PROTECTED] Date: Tue, 20 Mar 2007 09:49:34 +0100 Hi, Let's say you have an Asterisk server running. Which parameters would you check to improve service continuity ? I was thinking of : - telco lines status (make sure every is up) - registered hardphones If you use VoIP, add data network status (and possibly quality). Yuan Liu - config files backup (compare live and saved configuration files, if files differ, notifies the administration team) - systems variables (disk and CPU) - log files (trigger an alarm for every ERROR or NOTICE message in full logs) What do you think ? Regards ___ --Bandwidth 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] Zapateller not playing audio via SIP Trunk?
exten = i,1,Zapateller() Same happens if I use PlayTones(info) instead of ZapaTeller(). Same happens if I use Progress() before ZapaTeller or Playtones. Mit freundlichen Grüssen Benoit Panizzon -- I m p r o W a r e A G-System Services __ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 PrattelnFax +41 61 826 93 01 Schweiz Web http://www.imp.ch __ pgpQrWdA1exvv.pgp Description: PGP signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Configuring Faxs any help :)
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of younss azzayani Sent: Tuesday, March 20, 2007 3:52 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Configuring Faxs any help :) and this is the /var/spool/hylafax/log/c1: http://pastebin.ca/403282 cat /var/spool/hylafax/log/c3 :: http://pastebin.ca/403291 thank you Although there are errors, I do not think this is your problem. Maybe there is a better debug in /var/log/messages. Thanks, Steve Totaro http://www.asteriskhelpdesk.com KB3OPB ___ --Bandwidth 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] PROGRESS code
I have a PRI switch type national
Asterisk 1.2.16
Zaptel 1.2.15
If I call an invalid number I get
* PROGRESS with cause code 28 received
Asterisk continues to attempt to connect the call until the timeout is
reached and I hear ringing.
I want to capture the progress code, which I thought was in HANGUPCAUSE
but when I NoOp that variable it's always 16 when I dial an invalid
number...not 28
Also, I don't see how to immediately indicate the number is invalid,
without waiting for the channel to automatically hang up.
Is that just the way it works...I gotta wait?
Why is HANGUPCAUSE 16 but I get Progress cause code 28? 28 is clearly
correct because 11 is an invalid number format.
exten = _.,1,Dial(Zap/g1/${EXTEN})
exten = _.,n,NoOp(Dial Status is ${DIALSTATUS})
exten = _.,n,NoOp(Hang Up Clause is ${HANGUPCAUSE})
exten = _.,n,Congestion
-- Executing Dial(SIP/x.x.x.x-090bec30, Zap/g1/11) in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called g1/11
-- Zap/1-1 is proceeding passing it to SIP/x.x.x.x-090bec30
-- PROGRESS with cause code 28 received
-- Zap/1-1 is making progress passing it to SIP/x.x.x.x-090bec30
-- Channel 0/1, span 1 got hangup request
-- Hungup 'Zap/1-1'
== No one is available to answer at this time (1:0/0/0)
-- Executing NoOp(SIP/x.x.x.x-090bec30, Dial Status is NOANSWER)
in new stack
-- Executing NoOp(SIP/x.x.x.x-090bec30, Hang Up Cause is 16) in
new stack
-- Executing Congestion(SIP/x.x.x.x-090bec30, ) in new stack
== Spawn extension (pri-only, 11, 4) exited non-zero on
'SIP/x.x.x.x-090bec30'
-- Executing Hangup(SIP/x.x.x.x-090bec30, ) in new stack
Bill
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[asterisk-users] error, install freePbx
Hi, i try install FreePbx by tuturial in http://www.voip-info.org/wiki/view/Instalaci%C3%B3n+de+Asterisk+en+Ubuntu+Server+Paso+a+Pasoview_comment_id=13443 but i have this error when i try install freepbx: #pear install db No releases available for package pear.php.net/db Cannot initialize 'db' , invalid or missing package files Package db is not valid install failed Why this error? help me, please. -- Carlos Jerónimo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem with ATT Maintenance protocol in PRI connection, no B+D channels available
Hi guys, We are experiencing a problem with a T1 PRI connection. After trying a number of variations in the configuration files, the behavior is always the same: no B channels come up and the D channel doesn't appear to be working well. We can see there are ATT Maintenance messages being exchanged by asterisk and the provider, CONNECT and CONNECT ACKNOWLEDGE, but that doesn't appear to be enough to bring the D and B channels properly up. Are there any messages missing? When we attempt to make a call, we can see the Q.931 SETUP message being sent. But shortly after we are getting a LAPD DISC message, which ends up originating a Q.931 DISCONNECT message, terminating the call. What could be the problem here? Could there be any configuration issue on our side? Does libpri provide complete support to the ATT Maintenance protocol or could this connection be incompatible? Any help would be highly appreciated. Many thanks in advance, Paulo PS: Configuration files, messages and pri debug snippets follow zaptel.conf loadzone = us defaultzone=us #Span 1: TE2/0/1 T2XXP (PCI) Card 0 Span 1 PRI_T1 span=1,0,0,esf,b8zs,crc4 bchan=1-23 dchan=24 zapata.conf [channels] group = 0 usecallingpres = yes switchtype = national context = inbound signalling = pri_cpe usecallerid = yes channel = 1-23 messages -- Mar 19 15:32:23 NOTICE[3306] cdr.c: CDR logging disabled, data will be lost. Mar 19 15:32:23 WARNING[3306] pbx_ael.c: Unable to open '/etc/asterisk/extensions.ael': No such file or directory Mar 19 15:32:23 WARNING[3306] pbx.c: Requested contexts didn't get merged Mar 19 15:33:17 WARNING[3322] chan_zap.c: No D-channels available! Using Primary channel 24 as D-channel anyway! Mar 19 15:33:58 WARNING[3322] chan_zap.c: No D-channels available! Using Primary channel 24 as D-channel anyway! Mar 19 15:33:58 WARNING[3366] app_dial.c: Unable to forward voice [...] pri debug span -- [ 00 01 0a 0a 03 01 00 07 01 01 c0 18 01 ac ] Informational frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 N(S): 005 0: 0 N(R): 005 P: 0 10 bytes of data -- Restarting T203 counter Stopping T_203 timer Starting T_200 timer Protocol Discriminator: ATT Maintenance (3) len=10 Call Ref: len= 1 (reference 0/0x0) (Originator) Message type: CONNECT (7) [01 01 c0] IE: Change Status (len = 3) [18 01 ac] Channel ID (len= 3) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 1 ChanSel: As indicated in following octets ] (...) [ 02 01 0a 0c 03 01 00 0f 01 01 c0 18 01 ac ] Informational frame: SAPI: 00 C/R: 1 EA: 0 TEI: 000EA: 1 N(S): 005 0: 0 N(R): 006 P: 0 10 bytes of data -- ACKing all packets from 5 to (but not including) 6 -- Since there was nothing left, stopping T200 counter -- Stopping T203 counter since we got an ACK -- Nothing left, starting T203 counter Protocol Discriminator: ATT Maintenance (3) len=10 Call Ref: len= 1 (reference 0/0x0) (Originator) Message type: CONNECT ACKNOWLEDGE (15) [01 01 c0] IE: Change Status (len = 3) [18 01 ac] Channel ID (len= 3) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 1 ChanSel: As indicated in following octets ] (...) Protocol Discriminator: Q.931 (8) len=40 Call Ref: len= 2 (reference 2/0x2) (Originator) Message type: SETUP (5) [04 03 80 90 a2] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: u-Law (34) [18 03 a9 83 82] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 2 ] [1e 02 80 83] Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: User (0) Ext: 1 Progress Description: Calling equipment is non-ISDN. (3) ] [6c 06 21 80 37 31 30 30] Calling Number (len= 8) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation permitted, user number not screened (0) '7100' ] [70 0b a1 35 38 35 34 31 39 37 39 39 35] Called Number (len=13) [ Ext: 1 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '5854197995' ] (...) [ 02 01 53 ] Unnumbered frame: SAPI: 00 C/R: 1 EA: 0 TEI: 000EA: 1 M3: 2 P/F: 1 M2: 0 11: 3 [ DISC (disconnect) ] 0 bytes of data -- Got Disconnect from peer. Sending Unnumbered Acknowledgement [ 02 01 73 ] Unnumbered frame:
Re: [asterisk-users] Which parameters of a live Asterisk server would you monitor ?
On 3/20/07, Olivier [EMAIL PROTECTED] wrote: Let's say you have an Asterisk server running. Which parameters would you check to improve service continuity ? The tools I tend to use are vmstat, iftop (all VoIP, all the time), show registry and df. -HJC ___ --Bandwidth 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] Configuring Faxs any help :)
thank you :) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] which spandsp for asterisk 1.2.16 (eom)
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Re: [asterisk-users] error, install freePbx
perhaps you should try pear install DB However note that this mailing list has nothing to do with pear. Hi, i try install FreePbx by tuturial in http://www.voip-info.org/wiki/view/Instalaci%C3%B3n+de+Asterisk+en+Ubuntu+Server+Paso+a+Pasoview_comment_id=13443 but i have this error when i try install freepbx: #pear install db No releases available for package pear.php.net/db Cannot initialize 'db' , invalid or missing package files Package db is not valid install failed Why this error? help me, please. ___ --Bandwidth 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] Microsoft launches first PABX
Christopher, welcom to Vista, it's now possible. On 3/20/07, Christopher Chan [EMAIL PROTECTED] wrote: C F wrote: I think yes, why you disagree? Has Microsoft actually ever come with such useful features? It would be great to demonstrate the complete instability/insecurity of Windows based servers by have it shut down automatically in front the boss with a recorded message :D. Even better if it comes with a BSOD command :) ___ --Bandwidth 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] Configuring Faxs any help :)
younss azzayani wrote: and this is the /var/spool/hylafax/log/c1: http://pastebin.ca/403282 cat /var/spool/hylafax/log/c3 :: http://pastebin.ca/403291 What does zttest say? If it's below 99.98% then hardware configuration is where the problem is. Lee. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Microsoft launches first PABX
Quoting C F [EMAIL PROTECTED]: Awesome, the first PABX virus is just around the corner now that M$ has some bait for it to infect. In a world without borders we don't need windows or gates. Christopher, welcom to Vista, it's now possible. On 3/20/07, Christopher Chan [EMAIL PROTECTED] wrote: C F wrote: I think yes, why you disagree? Has Microsoft actually ever come with such useful features? It would be great to demonstrate the complete instability/insecurity of Windows based servers by have it shut down automatically in front the boss with a recorded message :D. Even better if it comes with a BSOD command :) ___ --Bandwidth 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 Jon Pounder _/_/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/ _/ _/_/_/ _/ _/_/ _/_/ _/_/ _/ _/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/_/ _/_/_/_/ Inline Internet Systems Inc. Thorold, Ontario, Canada Tools to Power Your e-Business Solutions www.inline.net www.ihtml.com www.ihtmlmerchant.com www.opayc.com This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with ATT Maintenance protocol in PRI connection, no B+D channels available
d-channel is in midle bchan=1-15,17-31 dchan=16 loadzone = it defaultzone = it Kanelbullar wrote: Hi guys, We are experiencing a problem with a T1 PRI connection. After trying a number of variations in the configuration files, the behavior is always the same: no B channels come up and the D channel doesn't appear to be working well. We can see there are ATT Maintenance messages being exchanged by asterisk and the provider, CONNECT and CONNECT ACKNOWLEDGE, but that doesn't appear to be enough to bring the D and B channels properly up. Are there any messages missing? When we attempt to make a call, we can see the Q.931 SETUP message being sent. But shortly after we are getting a LAPD DISC message, which ends up originating a Q.931 DISCONNECT message, terminating the call. What could be the problem here? * Could there be any configuration issue on our side? * Does libpri provide complete support to the ATT Maintenance protocol or could this connection be incompatible? Any help would be highly appreciated. Many thanks in advance, Paulo PS: Configuration files, messages and pri debug snippets follow zaptel.conf loadzone = us defaultzone=us #Span 1: TE2/0/1 T2XXP (PCI) Card 0 Span 1 PRI_T1 span=1,0,0,esf,b8zs,crc4 bchan=1-23 dchan=24 zapata.conf [channels] group = 0 usecallingpres = yes switchtype = national context = inbound signalling = pri_cpe usecallerid = yes channel = 1-23 messages -- Mar 19 15:32:23 NOTICE[3306] cdr.c: CDR logging disabled, data will be lost. Mar 19 15:32:23 WARNING[3306] pbx_ael.c: Unable to open '/etc/asterisk/extensions.ael': No such file or directory Mar 19 15:32:23 WARNING[3306] pbx.c: Requested contexts didn't get merged Mar 19 15:33:17 WARNING[3322] chan_zap.c: No D-channels available! Using Primary channel 24 as D-channel anyway! Mar 19 15:33:58 WARNING[3322] chan_zap.c: No D-channels available! Using Primary channel 24 as D-channel anyway! Mar 19 15:33:58 WARNING[3366] app_dial.c: Unable to forward voice [...] pri debug span -- [ 00 01 0a 0a 03 01 00 07 01 01 c0 18 01 ac ] Informational frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 N(S): 005 0: 0 N(R): 005 P: 0 10 bytes of data -- Restarting T203 counter Stopping T_203 timer Starting T_200 timer Protocol Discriminator: ATT Maintenance (3) len=10 Call Ref: len= 1 (reference 0/0x0) (Originator) Message type: CONNECT (7) [01 01 c0] IE: Change Status (len = 3) [18 01 ac] Channel ID (len= 3) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 1 ChanSel: As indicated in following octets ] (...) [ 02 01 0a 0c 03 01 00 0f 01 01 c0 18 01 ac ] Informational frame: SAPI: 00 C/R: 1 EA: 0 TEI: 000EA: 1 N(S): 005 0: 0 N(R): 006 P: 0 10 bytes of data -- ACKing all packets from 5 to (but not including) 6 -- Since there was nothing left, stopping T200 counter -- Stopping T203 counter since we got an ACK -- Nothing left, starting T203 counter Protocol Discriminator: ATT Maintenance (3) len=10 Call Ref: len= 1 (reference 0/0x0) (Originator) Message type: CONNECT ACKNOWLEDGE (15) [01 01 c0] IE: Change Status (len = 3) [18 01 ac] Channel ID (len= 3) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 1 ChanSel: As indicated in following octets ] (...) Protocol Discriminator: Q.931 (8) len=40 Call Ref: len= 2 (reference 2/0x2) (Originator) Message type: SETUP (5) [04 03 80 90 a2] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: u-Law (34) [18 03 a9 83 82] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 2 ] [1e 02 80 83] Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: User (0) Ext: 1 Progress Description: Calling equipment is non-ISDN. (3) ] [6c 06 21 80 37 31 30 30] Calling Number (len= 8) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation permitted, user number not screened (0) '7100' ] [70 0b a1 35 38 35 34 31 39 37 39 39 35] Called Number (len=13) [ Ext: 1 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '5854197995' ] (...) [ 02 01 53 ] Unnumbered frame: SAPI: 00 C/R: 1 EA: 0 TEI: 000EA: 1 M3: 2 P/F: 1 M2: 0 11: 3 [ DISC (disconnect) ] 0 bytes of data -- Got Disconnect from peer.
Re: [asterisk-users] transfer=mediaonly : can't hear nothing
Simone Cittadini wrote: I post once again here, sorry for the verbosity, if then in your opinion there's still something wrong with * internals and not with my understanding of the configs I'll open the bug. I would encourage you to open the bug anyway; I am currently at a trade show in California and will have limited ability to assist you in the next week, and nobody else has jumped into this thread to help :-) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with ATT Maintenance protocol in PRI connection, no B+D channels available
Thanks for your answer, Bruno. However, the configuration you provided is for an E1 connection and we are using a T1, having channel 23 as D channel. Bruno De Luca [EMAIL PROTECTED] escreveu: d-channel is in midle bchan=1-15,17-31 dchan=16 loadzone = it defaultzone = it Kanelbullar wrote: Hi guys, We are experiencing a problem with a T1 PRI connection. After trying a number of variations in the configuration files, the behavior is always the same: no B channels come up and the D channel doesn't appear to be working well. We can see there are ATT Maintenance messages being exchanged by asterisk and the provider, CONNECT and CONNECT ACKNOWLEDGE, but that doesn't appear to be enough to bring the D and B channels properly up. Are there any messages missing? When we attempt to make a call, we can see the Q.931 SETUP message being sent. But shortly after we are getting a LAPD DISC message, which ends up originating a Q.931 DISCONNECT message, terminating the call. What could be the problem here? Could there be any configuration issue on our side? Does libpri provide complete support to the ATT Maintenance protocol or could this connection be incompatible? Any help would be highly appreciated. Many thanks in advance, Paulo PS: Configuration files, messages and pri debug snippets follow zaptel.conf loadzone = us defaultzone=us #Span 1: TE2/0/1 T2XXP (PCI) Card 0 Span 1 PRI_T1 span=1,0,0,esf,b8zs,crc4 bchan=1-23 dchan=24 zapata.conf [channels] group = 0 usecallingpres = yes switchtype = national context = inbound signalling = pri_cpe usecallerid = yes channel = 1-23 messages -- Mar 19 15:32:23 NOTICE[3306] cdr.c: CDR logging disabled, data will be lost. Mar 19 15:32:23 WARNING[3306] pbx_ael.c: Unable to open '/etc/asterisk/extensions.ael': No such file or directory Mar 19 15:32:23 WARNING[3306] pbx.c: Requested contexts didn't get merged Mar 19 15:33:17 WARNING[3322] chan_zap.c: No D-channels available! Using Primary channel 24 as D-channel anyway! Mar 19 15:33:58 WARNING[3322] chan_zap.c: No D-channels available! Using Primary channel 24 as D-channel anyway! Mar 19 15:33:58 WARNING[3366] app_dial.c: Unable to forward voice [...] pri debug span -- [ 00 01 0a 0a 03 01 00 07 01 01 c0 18 01 ac ] Informational frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 N(S): 005 0: 0 N(R): 005 P: 0 10 bytes of data -- Restarting T203 counter Stopping T_203 timer Starting T_200 timer Protocol Discriminator: ATT Maintenance (3) len=10 Call Ref: len= 1 (reference 0/0x0) (Originator) Message type: CONNECT (7) [01 01 c0] IE: Change Status (len = 3) [18 01 ac] Channel ID (len= 3) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 1 ChanSel: As indicated in following octets ] (...) [ 02 01 0a 0c 03 01 00 0f 01 01 c0 18 01 ac ] Informational frame: SAPI: 00 C/R: 1 EA: 0 TEI: 000EA: 1 N(S): 005 0: 0 N(R): 006 P: 0 10 bytes of data -- ACKing all packets from 5 to (but not including) 6 -- Since there was nothing left, stopping T200 counter -- Stopping T203 counter since we got an ACK -- Nothing left, starting T203 counter Protocol Discriminator: ATT Maintenance (3) len=10 Call Ref: len= 1 (reference 0/0x0) (Originator) Message type: CONNECT ACKNOWLEDGE (15) [01 01 c0] IE: Change Status (len = 3) [18 01 ac] Channel ID (len= 3) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 1 ChanSel: As indicated in following octets ] (...) Protocol Discriminator: Q.931 (8) len=40 Call Ref: len= 2 (reference 2/0x2) (Originator) Message type: SETUP (5) [04 03 80 90 a2] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: u-Law (34) [18 03 a9 83 82] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 2 ] [1e 02 80 83] Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: User (0) Ext: 1 Progress Description: Calling equipment is non-ISDN. (3) ] [6c 06 21 80 37 31 30 30] Calling Number (len= 8) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation permitted, user number not screened (0) '7100' ] [70 0b a1 35 38 35 34 31 39 37 39 39 35] Called Number (len=13) [ Ext: 1 TON: National Number (2) NPI: ISDN/Telephony
RES: [asterisk-users] Need comparison between PBXtra, Trixbox, Thirdlane, Druid, Aheeva etc.
Hi! Could you please tell me why have you chosen the CentOS instead of any other Linux distribution? -- -Mensagem original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Em nome de shadowym Enviada em: sábado, 3 de março de 2007 14:17 Para: 'Asterisk Users Mailing List - Non-Commercial Discussion' Assunto: RE: [asterisk-users] Need comparison between PBXtra, Trixbox,Thirdlane, Druid, Aheeva etc. I'll second that, CentOS 4.4 + FreePBX 2.1.3 + Asterisk 1.2.13 + Sangoma A200D + Aastra 9133i's running 4 months without a reboot and no memory leaks fielding about 150 calls a day. Everyone loves the system. These are normal users used to tradtitional phone systems. I would not go as far as saying it's guaranteed solid as a rock until can get at least 1 year of uptime but it's pretty stable for sure! As far as the other GUI's. They all have their strengths. I would not rely on others opinions too much as everyones requirements and preferences seem to be a bit different. I would set up a test system and see if FreePBX works for you first. If not then you can start to explore some of the commecial alternatives. I would stay away from Trixbox. Besides, if you don't know enough Linux to set it all up yourself you should not be doing production installs IMHO. -Original Message- From: Mailing Lists [mailto:[EMAIL PROTECTED] Sent: Friday, March 02, 2007 6:32 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Need comparison between PBXtra, Trixbox,Thirdlane, Druid, Aheeva etc. On Mar 2, 2007, at 9:14 PM, Zeeshan Zakaria wrote: Hi, For a customer, I am looking for a good and reliable Asterisk based system. Five servers will be installed at different locations and will be linked together with each other. This system will work as a call center as well. It has to be a stable and reliable. Customer also needs GUIs for system administration and agents call activities. He also wants video conferencing Please help me select a good system. Thanks -- Zeeshan A Zakaria ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Running Asterisk with FreePBX on CentOS works great. I started with Trixbox and used it for a few weeks before I simply downloaded and compiled my own. Honestly, it is not that hard. Just follow the instructions. I created a series of scripts to run against a fresh CentOS install which deal with compiling and installing everything, including the FreePBX dependencies. I'd be happy to share it. I don't do any video conferencing, and I don't patch Asterisk for faxing. FreePBX works quite well. Combined with all the features of modern SIP phones, there is nothing you can't do. I run my systems using the Intel 975XBX2 motherboard (975 chipset), which I assembled by buying components from New Egg. Very stable - no issues with CentOS. -Joe ___ --Bandwidth 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] asterisk on debian
hello friends, I want to install Asterisk on a Debian machine. I need to download the sources or just with apt-get install is enought??? 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
[asterisk-users] modem passthru
Our setup is: 9.6k Modem -analog- Mitel SX-200 -(pri)- Asterisk -(pri) - Telco The modem works fine with the Mitel directly connected to the Telco, but once we add Asterisk in between connections start failing. I suspect the issue is caused by the echo canceller, since I believe the issue appear about the time we turned echo cancellation on (for the IAX users). We don't need echo cancellation for PRI to PRI calls. I've looked around, but I am finding conflicting opinions on what the echocancelwhenbridged line does. Some say it turns off the echo canceller for TDM to TDM calls if set to yes, some say if it is set to no. Which is correct? ;Zaptel Channels Configurations (zapata.conf) [trunkgroups] [channels] context=default usecallerid=yes facilityenable=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes resetinterval=never rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no ;Sangoma A102 port 1 [slot:10 bus:1 span: 1] switchtype=national context=from-internal group=1 signalling=pri_net channel = 1-23 ;Sangoma A102 port 2 [slot:10 bus:1 span: 2] switchtype=national context=from-pstn group=0 signalling=pri_cpe channel = 25-47 ___ --Bandwidth 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] GXP-2000 Phones with Cisco 3560 PoE Switch
We are connecting the GXP2000 to a Cisco POE switch 3650, it the default mode at 15.4 for each phone the phones power up but we can only have 24 on a 48 port switch, when we adjust the setting to 7000 (which is what we calculate the phone to use) they don't power up ... we tried all the power settings from 7000 up to 154000 but it only powers up at 15.4, we should be able to get almost 48 on the POE switch with 370 watts of power on it. Any trick or adjustments you can think we need to make. Cory Andrews ___ --Bandwidth 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] Zaptel 1.2.16 Released
The Asterisk and Zaptel development teams have released Zaptel version 1.2.16. In addition to minor bug fixes, this release fixes a build-time problem on systems where the default language is not English, and also corrects a regression in the driver for the Digium dual- and quad-span cards with hardware echo cancelers that could result in kernel panics. Thanks for your support of Asterisk and Zaptel! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with ATT Maintenance protocol in PRI connection, no B+D channels available
Kanelbullar wrote: Hi guys, We are experiencing a problem with a T1 PRI connection. After trying a number of variations in the configuration files, the behavior is always the same: no B channels come up and the D channel doesn't appear to be working well. We can see there are ATT Maintenance messages being exchanged by asterisk and the provider, CONNECT and CONNECT ACKNOWLEDGE, but that doesn't appear to be enough to bring the D and B channels properly up. Are there any messages missing? When we attempt to make a call, we can see the Q.931 SETUP message being sent. But shortly after we are getting a LAPD DISC message, which ends up originating a Q.931 DISCONNECT message, terminating the call. What could be the problem here? * Could there be any configuration issue on our side? * Does libpri provide complete support to the ATT Maintenance protocol or could this connection be incompatible? span=1,0,0,esf,b8zs,crc4 This needs to be span=1,1,0,esf,b8zs I'm not sure if the crc4 is necessary. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth 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 on debian
On 3/20/07, Josu Lazkano Lete [EMAIL PROTECTED] wrote: hello friends, I want to install Asterisk on a Debian machine. I need to download the sources or just with apt-get install is enought??? Depends on the version you want to install. You can install with apt-get install asterisk, of course. More info here: http://www.voip-info.org/tiki-index.php?page=Asterisk+Linux+Debian Regards, Victor ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk on debian
On 3/20/07, Josu Lazkano Lete [EMAIL PROTECTED] wrote: I want to install Asterisk on a Debian machine. I need to download the sources or just with apt-get install is enought??? It depends on what version do you want to use. In sarge is only the version 1.0.7. In etch is 1.2.13, but the 1.2 branch is at 1.2.16 and there's the version 1.4.1 is out also. ___ --Bandwidth 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 on debian
Josu Lazkano Lete wrote: I need to download the sources or just with apt-get install is enought??? apt-get is the easiest way, but won't give you the latest release. ___ --Bandwidth 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] Configuring Faxs any help :)
Hello, everyone. I get Best: 99.975586 -- Worst: 99.975586 -- Average: 99.975586 with zttest Where do I have to start looking for hardware errors? Thanks in advance younss azzayani wrote: and this is the /var/spool/hylafax/log/c1: http://pastebin.ca/403282 cat /var/spool/hylafax/log/c3 :: http://pastebin.ca/403291 What does zttest say? If it's below 99.98% then hardware configuration is where the problem is. Lee. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth 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 on debian
You could download the source from asterisk.org and follow the install instructions. You could also use SVN to download the source. Also, there are a few binary package links found at http://www.voip-info.org/wiki/index.php?page=Asterisk+Download. Bobby Crawford _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Josu Lazkano Lete Sent: Tuesday, March 20, 2007 8:58 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] asterisk on debian hello friends, I want to install Asterisk on a Debian machine. I need to download the sources or just with apt-get install is enought??? thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] modem passthru
On 3/20/07, Mark Farver [EMAIL PROTECTED] wrote: I suspect the issue is caused by the echo canceller, since I believe the issue appear about the time we turned echo cancellation on (for the IAX users). We don't need echo cancellation for PRI to PRI calls. I've looked around, but I am finding conflicting opinions on what the echocancelwhenbridged line does. Some say it turns off the echo canceller for TDM to TDM calls if set to yes, some say if it is set to no. Which is correct? Mark, The name of the option is fairly clear. If you want echo cancellation even when the call is bridged directly from card to card, set the option to 'yes'. Otherwise, set the option to 'no'. This option should be 'no' in the majority of cases. William ___ --Bandwidth 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] Activating Incoming Demo
Hello, What numbers do I dial to make an analog phone attached to an TDM400P ring via the asterisk demo after installation and starting asterisks? Thanks, Ed ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: qozap: t3 timer expired for span ...
This problem / messages has not gone away ... No one's got any ideas or explanation about what the card is trying to tell me? -- Chris Chris Earle [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] bristuff-0.2.0-RC8s two isdn lines plugged into first two ports and like I said, also a digium tdm400 card in there for analog phones this 'timer' error message it is something to do with the qozap driver isn't it? not sure Thanks for any ideas! -- Chris Tzafrir Cohen [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] On Thu, Mar 15, 2007 at 10:30:23AM -0500, Chris Earle (CBL) wrote: Hi all message: qozap: t4 timer expired for span 2 qozap: t4 timer expired for span 3 qozap: t3 timer expired for span 2 qozap t3 timer expired for span 3 Which version is it of bristuff? wow -- what does this mean!? all of a sudden showing up on my server ... no change after reboot .. Junghanns QuadBRI card in place Anything connected to it? Where exactly? affecting outgoing faxing?! (between bridged TDM400 analog card and QuadBRI) Not a clue why this is .. incoming/outgoing voice calls work, incoming faxes even work but when outgoing fax is dialed, says no one is availale to answer at this time The error has not ever been there before and as far as I know, no isdn wiring has been changed or anything ideas, appreciated! -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth 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] asterisk on debian
Hi Josu, I've done it both ways, and they both generally work equally well (so long as the package maintainers are doing a decent job). As Victor mentioned though, the version you wish to install plays a factor in this. I found the Asterisk build in the repos to be a bit out dated. Also, it's always bothered me having to wait on another party to create a package so that I can fix a security vulnerability. I've just gone with the straight from source method for now, but that's all personal opinion on that matter really. The bottom line is that if you want the latest and greatest (in terms of both feature sets and security updates), build it yourself. Apt-get may be easier, but there's plenty of good guides to get you going with building from source. Alex On 3/20/07, Josu Lazkano Lete [EMAIL PROTECTED] wrote: hello friends, I want to install Asterisk on a Debian machine. I need to download the sources or just with apt-get install is enought??? 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 -- Alex Robar [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Configuring Faxs any help :)
zfter running zttool i got: --- Results after 303 passes --- Best: 100.00 -- Worst: 99.987793 -- Average: 99.994914 ___ --Bandwidth 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] Configuring Faxs any help :)
dima wrote: I get Best: 99.975586 -- Worst: 99.975586 -- Average: 99.975586 with zttest Where do I have to start looking for hardware errors? I would start with IRQ sharing. Make sure that your Zap hardware isn't sharing an IRQ. Secondly, you want to have it with a fairly high priority, probably before your network card and your hard drive interfaces (meaning if your NIC is on IRQ 11 then *don't* put your Zap on IRQ 12. It should be on 9 or 10 or something.) Lee. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Configuring Faxs any help :)
younss azzayani wrote: zfter running zttool i got: --- Results after 303 passes --- Best: 100.00 -- Worst: 99.987793 -- Average: 99.994914 Okay. Then at this point you need to put record in your iaxmodem config file, restart iaxmodem, attempt the call again, and afterwards send me the *.raw files that should now be in /tmp/. (Or follow the instructions in README to see how to make use of them.) Lee. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Conference server (or how to make a call with more than 3 u
This can be done like this:
;user extensions
exten= 1,1,Dial(SIP/U1,,Tt)
exten= 2,1,Dial(SIP/U2,,Tt)
exten= 3,1,Dial(SIP/U2,,Tt)
;secretary extensions
exten= 4,1,Dial(SIP/Secretary,Tt)
the Tt option in dialplan lets the secretary to transfer the user
;conference extensions
exten= 123,1,Meetme(${EXTEN})
exten= 234,1,Meetme(${EXTEN})
your secratary dials any user who she wants to join the boss in a conference
room. after user answeres your secratary presses # button (or transfer
combination keys defined in features.conf) she will hear a transfer message
followed by a dialtone, here she has to dial conference room no where she
wants to throw the user to join the conference like 123, or 234.
Hope its helpfull
On 3/20/07, Angel Heart [EMAIL PROTECTED] wrote:
Hi Yehavi,
Yes, this can be done. We are currently using this features. The
Secretaries making the calls to who ever her Boss wants to join the
conference she then just transfer the calls into the conference room. You
can even annouce the name of the newly arrived calls in the conference.
Like; Mr. Mateevitsi join the conference or Mr. Mateevitsi leaved the
conference if one's leave the conference. I had created one coference room
for every department.
Regards.
Angel
*Victor Mateevitsi [EMAIL PROTECTED]* wrote:
Or, you can just transfer the calls into the conference room.
On 3/19/07, Lee Jenkins [EMAIL PROTECTED] wrote:
Yehavi Bourvine +972-8-9489444 wrote:
Why not use the MeetMe feature of asterisk?
I need the person who initiated the conference call to call the others
and join
them by herself. If I understand correctly, with the MeetMe you have
to
initialize the conference and then people dial by themselves into it.
This
won't be acceptable by the secretaries here...
Yehavi,
Can you make a script that uses call files to get everyone into the
conference?
--
Warm Regards,
Lee
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Don't get soaked. Take a quick peek at the forecast
with theYahoo! Search weather shortcut.
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Regards
Rizwan Hisham
Software Engineer
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[asterisk-users] Asterisk Automated Outbound Messaging
I have a client application looking for an Asterisk based solution. Client wants to deliver pre-recorded messages for a variety of clients. Wondering if anyone is offering an middleware for Asterisk for management of outbound messaging? Email me. Thanks Cory Andrews ___ --Bandwidth 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] Microsoft launches first PABX
If anyone is at VON I would be curious to know what sort of hardware they are using. Quanta Computers-booth 1445 Dlink, Uniden apparently decided not to attend according to the exhibitors list From the back of the boxes it looks like ITX but maybe not ix86 CPU. Nothing to indicate how they plan to connect to the PSTN. Maybe 3rd party gateways? One reviewer was quick to point out microsoft has dabbled in this before with their cordless phone with TAPI integration to Outlook and voice recognition. It quietly went away. http://www.tmcnet.com/articles/ctimag/0399/0399micro.htm -Original Message- From: James Andrewartha [mailto:[EMAIL PROTECTED] Sent: Tuesday, March 20, 2007 12:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Microsoft launches first PABX Christopher Chan wrote: C F wrote: I think yes, why you disagree? Has Microsoft actually ever come with such useful features? It would be great to demonstrate the complete instability/insecurity of Windows based servers by have it shut down automatically in front the boss with a recorded message :D. Even better if it comes with a BSOD command :) It's doable with Vista, but you need voice recognition turned on first: http://blogs.zdnet.com/Ou/?p=416 -- James Andrewartha ___ --Bandwidth 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] High Pitched Noise
Question: After about having the server running for about an hour, our callers occationally hear a high pitched beep that lasts the entire call. In some cases, the noise doesn't start until a minute or 2 into the call, while others last the entire call. In some of the more serious cases, calls are dropped after the noise has occurred as well. Another symptom has been really bad static on a specific channel. After reseating the card to try to fix both this as well as the problem above, the problem usually goes away, but it seems to come back quicker each time. Also, the channel that the static occurs on changes after each reseating (after some time). Could this just be a bad digium card? Or could it be a bad PCI bus and we should try a new PC? Doesn't seem like an asterisk problem, but I won't rule it out until I know what it is. Any thoughts? Rob ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Problem with ATT Maintenance protocol inPRI connection, no B+D channels available
span=1,0,0,esf,b8zs,crc4 This needs to be span=1,1,0,esf,b8zs I'm not sure if the crc4 is necessary. Doug I concur with Doug. I have two PRI's in one system. My zaptel.conf looks like this: span=1,1,0,esf,b8zs # PRI line - LD Qwest (interstate) bchan=1-23 dchan=24 span=2,2,0,esf,b8zs # PRI line - SBCLD (intrastate/local) bchan=25-47 dchan=48 HTH, MC ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Can't Compile w/HPEC
Hi All - I've been trying to compile Zaptel w/ HPEC, but I've been unsuccessful. The system is CentOS 4.4, zaptel version 1.2.15. I believe I've got all the requisite files, and they're in the right locations in the zaptel tree. When I compile, I get the following warning from make: Warning: could not find /usr/src/zaptel-1.2.15/hpec/.hpec_x86_32.o.cmd for /usr/src/zaptel-1.2.15/hpec/hpec_x86_32.o It's only a warning, so I went ahead with the process of registration and activation. Registration went fine, but when I try to activate the licenses, it gives the following message: Found valid HPEC licenses for 4 channels. The Zaptel module on this system appears to have been built without HPEC support. Please check your Zaptel build. So, apparently, the warning is more than a warning. I'm officially out of warranty on my TDM card, so I can't call Digium. Any help is appreciated. Thanks! Noah ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Can't Compile w/HPEC
And I just saw that Zaptel 1.2.16 is out. I'll give that a try... On 3/20/07, Noah Miller [EMAIL PROTECTED] wrote: Hi All - I've been trying to compile Zaptel w/ HPEC, but I've been unsuccessful. The system is CentOS 4.4, zaptel version 1.2.15. I believe I've got all the requisite files, and they're in the right locations in the zaptel tree. When I compile, I get the following warning from make: Warning: could not find /usr/src/zaptel-1.2.15/hpec/.hpec_x86_32.o.cmd for /usr/src/zaptel-1.2.15/hpec/hpec_x86_32.o It's only a warning, so I went ahead with the process of registration and activation. Registration went fine, but when I try to activate the licenses, it gives the following message: Found valid HPEC licenses for 4 channels. The Zaptel module on this system appears to have been built without HPEC support. Please check your Zaptel build. So, apparently, the warning is more than a warning. I'm officially out of warranty on my TDM card, so I can't call Digium. Any help is appreciated. Thanks! Noah ___ --Bandwidth 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] Configuring Faxs any help :)
Indeed, the IRQ priority of my card is low. Sorry for being lame, but does that have to be set in BIOS? Are there any best practices for choosing an IRQ or I can set it to any free number? I would start with IRQ sharing. Make sure that your Zap hardware isn't sharing an IRQ. Secondly, you want to have it with a fairly high priority, probably before your network card and your hard drive interfaces (meaning if your NIC is on IRQ 11 then *don't* put your Zap on IRQ 12. It should be on 9 or 10 or something.) Lee. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Can't Compile w/HPEC
And I just saw that Zaptel 1.2.16 is out. I'll give that a try... And no... the problem still exists with zaptel 1.2.16 On 3/20/07, Noah Miller [EMAIL PROTECTED] wrote: Hi All - I've been trying to compile Zaptel w/ HPEC, but I've been unsuccessful. The system is CentOS 4.4, zaptel version 1.2.15. I believe I've got all the requisite files, and they're in the right locations in the zaptel tree. When I compile, I get the following warning from make: Warning: could not find /usr/src/zaptel-1.2.15/hpec/.hpec_x86_32.o.cmd for /usr/src/zaptel-1.2.15/hpec/hpec_x86_32.o It's only a warning, so I went ahead with the process of registration and activation. Registration went fine, but when I try to activate the licenses, it gives the following message: Found valid HPEC licenses for 4 channels. The Zaptel module on this system appears to have been built without HPEC support. Please check your Zaptel build. So, apparently, the warning is more than a warning. I'm officially out of warranty on my TDM card, so I can't call Digium. Any help is appreciated. Thanks! Noah ___ --Bandwidth 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] Configuring Faxs any help :)
That is going to depend upon your motherboard hardware, most likely. Understand that in most cases the NIC and hard drives are usually going to be the most demanding interrupt resource competitors to the Zap hardware, and that the Zap hardware cannot often operate properly (i.e. for fax) being a second-class citizen to those. If you have some other hardware that is equally as demanding then you will need to also account for that. Generally you can adjust IRQs in the motherboard BIOS or by rearranging the PCI cards. Lee. dima wrote: Indeed, the IRQ priority of my card is low. Sorry for being lame, but does that have to be set in BIOS? Are there any "best practices" for choosing an IRQ or I can set it to any free number? I would start with IRQ sharing. Make sure that your Zap hardware isn't sharing an IRQ. Secondly, you want to have it with a fairly high priority, probably before your network card and your hard drive interfaces (meaning if your NIC is on IRQ 11 then *don't* put your Zap on IRQ 12. It should be on 9 or 10 or something.) Lee. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Can't Compile w/HPEC
And I just saw that Zaptel 1.2.16 is out. I'll give that a try... And no... the problem still exists with zaptel 1.2.16 Problem solved. The warning meant nothing, compile was fine. For some reason the old non-HPEC zaptel kernel module wouldn't unload. Now, why that was, I don't know. ___ --Bandwidth 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] codec_zap and Asterisk 1.4.1
I've downloaded: asterisk-1.4.1 zaptel-1.4.0 I've compiled and installed zaptel. When I go to install asterisk I do: ./configure make menuselect I then take a look under the codec selection menu and I see that codec_zap can not be compiled. * Asterisk Module Selection * Press 'h' for help. [*] 1. codec_adpcm [*] 2. codec_alaw [*] 3. codec_a_mu [*] 4. codec_g726 [*] 5. codec_gsm [*] 6. codec_ilbc [*] 7. codec_lpc10 XXX 8. codec_speex [*] 9. codec_ulaw XXX 10. codec_zap Generic Zaptel Transcoder Codec Translator Depends on: zaptel_transcode(E), zaptel(E) I see this in the changelog: 2007-02-24 00:53 + [r56548] Kevin P. Fleming [EMAIL PROTECTED] * codecs/codec_zap.c: update to match zaptel 1.4 API change that was committed a few minutes ago 2007-01-22 19:41 + [r51411] Russell Bryant [EMAIL PROTECTED] * /: Blocked revisions 51410 via svnmerge r51410 | russell | 2007-01-22 13:39:30 -0600 (Mon, 22 Jan 2007) | 3 lines Merge codec_zap support for the transcoder card. This is a standalone codec module so it will not affect anything else. 2007-01-05 23:16 + [r49705] Jason Parker [EMAIL PROTECTED] * channels/chan_zap.c, codecs/codec_zap.c: Make codec_zap and chan_zap also depend on zaptel. This fixes an issue (8727) with zaptel being in a different directory, using --with-zaptel. 2007-01-01 23:34 + [r49098-49102] Kevin P. Fleming [EMAIL PROTECTED] * channels/chan_zap.c, build_tools/menuselect-deps.in, configure, configure.ac, codecs/codec_zap.c: check specifically for VLDTMF and transcoding support in the system's Zaptel installation, and make only the modules that need those features dependent on them (this will allow building the other Zaptel-using parts of Asterisk against older versions of Zaptel or those on other platforms that haven't caught up yet to the Linux version) I'm a little confused. I'm running a sangoma a200 card in my server. I thought I needed codec_zap. Do I need to wait until zaptel-1.4.1 gets released to be able to compile this or do I just not need it all together? Any insight would be appreciated. Thanks in advance, Jeremiah ___ --Bandwidth 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] High Pitched Noise
Hi Rob - After about having the server running for about an hour, our callers occationally hear a high pitched beep that lasts the entire call. In some cases, the noise doesn't start until a minute or 2 into the call, while others last the entire call. In some of the more serious cases, calls are dropped after the noise has occurred as well. Another symptom has been really bad static on a specific channel. After reseating the card to try to fix both this as well as the problem above, the problem usually goes away, but it seems to come back quicker each time. Also, the channel that the static occurs on changes after each reseating (after some time). What kind of PSTN lines are they? If they're POTS lines, can you plug a regular phone in and test the noise then? Also have you looked at other hardware devices inside the asterisk box? I've heard disk drives (hard, floppy, optical) that make loud enough noises that they interfere with analog phone lines. Do you have another machine to test the card in? - Noah ___ --Bandwidth 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] DNIS/DNID
On 3/16/07, Trevor Peirce [EMAIL PROTECTED] wrote:
Mark Quitoriano wrote:
Hi i have an asterisk pbx with E1 port connected to another PBX. Im
trying to send the DNID/DNIS to the PBX here's my dialplan
exten = 888111,1,Dial(ZAP/g2)
exten = 888111,n,Hangup()
The PBX just get the number 2 as it's DNIS when i change it to ZAP/1
or ZAP/g1 the PBX get the number 1. What should i add to send the
extension number as DNID/DNIS?
exten = 888111,1,Dial(ZAP/g2/${EXTEN})
Right now you're trying to dial the number g2, instead of using group 2.
yes g2 where it means group 2 right? my DNIS setup is now working im using
this config.
exten = 888111,1,Dial(ZAP/g2/*${EXTEN}*)
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RE: [asterisk-users] Need comparison between PBXtra, Trixbox, Thirdlane, Druid, Aheeva etc.
The decision to use CentOS was(is) simple for me. That is the standard OS chosen by Asterisk and FreePBX developers more or less. At least it was in the early days. From there, the majority of people using Asterisk/FreePBX have chosen CentOS. So in a nutshell, it is the most used, most tested, most documented OS. That is not to say people have not had and cannot have great success with other distributions. For me it is what makes the most sense. Keep it simple and all that. Just my opinion. -Original Message- From: Alcides [mailto:[EMAIL PROTECTED] Sent: Tuesday, March 20, 2007 6:58 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RES: [asterisk-users] Need comparison between PBXtra, Trixbox,Thirdlane, Druid, Aheeva etc. Hi! Could you please tell me why have you chosen the CentOS instead of any other Linux distribution? -- -Mensagem original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Em nome de shadowym Enviada em: sábado, 3 de março de 2007 14:17 Para: 'Asterisk Users Mailing List - Non-Commercial Discussion' Assunto: RE: [asterisk-users] Need comparison between PBXtra, Trixbox,Thirdlane, Druid, Aheeva etc. I'll second that, CentOS 4.4 + FreePBX 2.1.3 + Asterisk 1.2.13 + Sangoma A200D + Aastra 9133i's running 4 months without a reboot and no memory leaks fielding about 150 calls a day. Everyone loves the system. These are normal users used to tradtitional phone systems. I would not go as far as saying it's guaranteed solid as a rock until can get at least 1 year of uptime but it's pretty stable for sure! As far as the other GUI's. They all have their strengths. I would not rely on others opinions too much as everyones requirements and preferences seem to be a bit different. I would set up a test system and see if FreePBX works for you first. If not then you can start to explore some of the commecial alternatives. I would stay away from Trixbox. Besides, if you don't know enough Linux to set it all up yourself you should not be doing production installs IMHO. -Original Message- From: Mailing Lists [mailto:[EMAIL PROTECTED] Sent: Friday, March 02, 2007 6:32 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Need comparison between PBXtra, Trixbox,Thirdlane, Druid, Aheeva etc. On Mar 2, 2007, at 9:14 PM, Zeeshan Zakaria wrote: Hi, For a customer, I am looking for a good and reliable Asterisk based system. Five servers will be installed at different locations and will be linked together with each other. This system will work as a call center as well. It has to be a stable and reliable. Customer also needs GUIs for system administration and agents call activities. He also wants video conferencing Please help me select a good system. Thanks -- Zeeshan A Zakaria ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Running Asterisk with FreePBX on CentOS works great. I started with Trixbox and used it for a few weeks before I simply downloaded and compiled my own. Honestly, it is not that hard. Just follow the instructions. I created a series of scripts to run against a fresh CentOS install which deal with compiling and installing everything, including the FreePBX dependencies. I'd be happy to share it. I don't do any video conferencing, and I don't patch Asterisk for faxing. FreePBX works quite well. Combined with all the features of modern SIP phones, there is nothing you can't do. I run my systems using the Intel 975XBX2 motherboard (975 chipset), which I assembled by buying components from New Egg. Very stable - no issues with CentOS. -Joe ___ --Bandwidth 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 Automated Outbound Messaging
Cory Andrews wrote: I have a client application looking for an Asterisk based solution. Client wants to deliver pre-recorded messages for a variety of clients. Wondering if anyone is offering an middleware for Asterisk for management of outbound messaging? Someone can correct me if I'm wrong, but I think a friend of mine mentioned that TrixBox has a gab cast function. It also shouldn't be that difficult to put together a script to do this. I actually have plans to do this myself, but no need for it just yet... -- Warm Regards, Lee ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] ExternalIVR() Dialplan function and Festival
I ended up using text2wave to create a wav file and then added it to the prompt list and that worked. Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Ruggles Sent: Monday, March 19, 2007 5:07 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] ExternalIVR() Dialplan function and Festival Is there any way to use Festival from script called by the ExternalIVR() dialplan function? Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth 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] modem passthru
William Moore wrote: The name of the option is fairly clear. If you want echo cancellation even when the call is bridged directly from card to card, set the option to 'yes'. Otherwise, set the option to 'no'. This option should be 'no' in the majority of cases. Thanks, that's what I thought. I found references online that said that the option actually behave opposite to the logical interpetation. Thanks for clearing it up. It turns out it did not actually fix my problem, so anyone have other pointers on how to troubleshoot this one? Modems are tricky, no feedback as to why the calls fails. Mark Farver ___ --Bandwidth 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 Automated Outbound Messaging
These folks have 6-8 T's worth of outbound they do on a daily basis, I need an interface that would allow them to stick a comma delimited file or file(s) in every day via FTP, the file would contain call #'s, and some additional variables, and then the Asterisk box would schedule the calls. It would pull a voice file locally and deliver to answering machines or live call recipients. Cory Andrews -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lee Jenkins Sent: Tuesday, March 20, 2007 3:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk Automated Outbound Messaging Cory Andrews wrote: I have a client application looking for an Asterisk based solution. Client wants to deliver pre-recorded messages for a variety of clients. Wondering if anyone is offering an middleware for Asterisk for management of outbound messaging? Someone can correct me if I'm wrong, but I think a friend of mine mentioned that TrixBox has a gab cast function. It also shouldn't be that difficult to put together a script to do this. I actually have plans to do this myself, but no need for it just yet... -- Warm Regards, Lee ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth 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] error, install freePbx
thanks, and sorry because the mailing list nothing to do with pear. but it'sa for install freePBX. pear install DB not works. more any sugestion? 2007/3/20, dima [EMAIL PROTECTED]: perhaps you should try pear install DB However note that this mailing list has nothing to do with pear. Hi, i try install FreePbx by tuturial in http://www.voip-info.org/wiki/view/Instalaci%C3%B3n+de+Asterisk+en+Ubuntu+Server+Paso+a+Pasoview_comment_id=13443 but i have this error when i try install freepbx: #pear install db No releases available for package pear.php.net/db Cannot initialize 'db' , invalid or missing package files Package db is not valid install failed Why this error? help me, please. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Carlos Jerónimo ___ --Bandwidth 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 Automated Outbound Messaging
It sounds like what you want is called a predictive dialer? There are several listed on the voip-info wiki. On 3/20/07, Lee Jenkins [EMAIL PROTECTED] wrote: Cory Andrews wrote: I have a client application looking for an Asterisk based solution. Client wants to deliver pre-recorded messages for a variety of clients. Wondering if anyone is offering an middleware for Asterisk for management of outbound messaging? Someone can correct me if I'm wrong, but I think a friend of mine mentioned that TrixBox has a gab cast function. It also shouldn't be that difficult to put together a script to do this. I actually have plans to do this myself, but no need for it just yet... -- Warm Regards, Lee ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mitchel Constantin Snap - A desktop user interface for Asterisk www.snapanumber.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] error, install freePbx
Hi Dima, You're better off following the Ubuntu guide written by the FreePBX developers: http://aussievoip.com.au/wiki/freePBX-Ubuntu Alex On 3/20/07, dima [EMAIL PROTECTED] wrote: perhaps you should try pear install DB However note that this mailing list has nothing to do with pear. Hi, i try install FreePbx by tuturial in http://www.voip-info.org/wiki/view/Instalaci%C3%B3n+de+Asterisk+en+Ubuntu+Server+Paso+a+Pasoview_comment_id=13443 but i have this error when i try install freepbx: #pear install db No releases available for package pear.php.net/db Cannot initialize 'db' , invalid or missing package files Package db is not valid install failed Why this error? help me, please. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Robar [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] High Pitched Noise
This is a PRI 24 channel line. We have backup pots lines, but they aren't in use. The problem we were having was happening on only a single channel or 2. Rob Noah Miller wrote: Hi Rob - After about having the server running for about an hour, our callers occationally hear a high pitched beep that lasts the entire call. In some cases, the noise doesn't start until a minute or 2 into the call, while others last the entire call. In some of the more serious cases, calls are dropped after the noise has occurred as well. Another symptom has been really bad static on a specific channel. After reseating the card to try to fix both this as well as the problem above, the problem usually goes away, but it seems to come back quicker each time. Also, the channel that the static occurs on changes after each reseating (after some time). What kind of PSTN lines are they? If they're POTS lines, can you plug a regular phone in and test the noise then? Also have you looked at other hardware devices inside the asterisk box? I've heard disk drives (hard, floppy, optical) that make loud enough noises that they interfere with analog phone lines. Do you have another machine to test the card in? - Noah ___ --Bandwidth 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 Automated Outbound Messaging
Cory Andrews wrote: These folks have 6-8 T's worth of outbound they do on a daily basis, I need an interface that would allow them to stick a comma delimited file or file(s) in every day via FTP, the file would contain call #'s, and some additional variables, and then the Asterisk box would schedule the calls. It would pull a voice file locally and deliver to answering machines or live call recipients. Cory Andrews -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lee Jenkins Sent: Tuesday, March 20, 2007 3:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk Automated Outbound Messaging Cory Andrews wrote: I have a client application looking for an Asterisk based solution. Client wants to deliver pre-recorded messages for a variety of clients. Wondering if anyone is offering an middleware for Asterisk for management of outbound messaging? Someone can correct me if I'm wrong, but I think a friend of mine mentioned that TrixBox has a gab cast function. It also shouldn't be that difficult to put together a script to do this. I actually have plans to do this myself, but no need for it just yet... If they want a decent interface to see the next caller before calling, you might want to have a database that reads in all the numbers, then users that grab the next non-checked number from the database. This also gives you the option of leaving notes with that call (such as calling back, etc). Then when ready, press the call button which creates a call file. Rob ___ --Bandwidth 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] Zaptel silly issue
Hi For starters, when you want to post a message to the list, don't just reply to an existing message. Start a new message. If you look at the archives of this list, you'll see your message as a reply to another message. This is because when you reply, the miler preserves some threading-related headers. See my reply in-line, On Mon, Mar 19, 2007 at 03:17:00PM -0500, Brad Sumrall wrote: I am geet this error, I assume because I have zero digium hardware installed. This is to be an entirely web based PBX. But you do have shell access, I hope. Can anyone point me to an easy 123 for installing zaptel in dummy form? I need music on hold for a VPS server. What type of VPS server is it? Do you have root access? Can you install load your own kernel modules? Which kernel version do you have? Which system is it? What error do you get? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth 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] High Pitched Noise
Could you be having ECFO? See: http://lists.digium.com/pipermail/asterisk-dev/2006-August/022062.html http://lists.digium.com/pipermail/asterisk-dev/2006-August/022111.html Rob Schall wrote: This is a PRI 24 channel line. We have backup pots lines, but they aren't in use. The problem we were having was happening on only a single channel or 2. Rob Noah Miller wrote: Hi Rob - After about having the server running for about an hour, our callers occationally hear a high pitched beep that lasts the entire call. In some cases, the noise doesn't start until a minute or 2 into the call, while others last the entire call. In some of the more serious cases, calls are dropped after the noise has occurred as well. Another symptom has been really bad static on a specific channel. After reseating the card to try to fix both this as well as the problem above, the problem usually goes away, but it seems to come back quicker each time. Also, the channel that the static occurs on changes after each reseating (after some time). What kind of PSTN lines are they? If they're POTS lines, can you plug a regular phone in and test the noise then? Also have you looked at other hardware devices inside the asterisk box? I've heard disk drives (hard, floppy, optical) that make loud enough noises that they interfere with analog phone lines. Do you have another machine to test the card in? ___ --Bandwidth 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 Automated Outbound Messaging
From: Rob Schall [EMAIL PROTECTED] Date: Tue, 20 Mar 2007 16:00:01 -0500 Cory Andrews wrote: These folks have 6-8 T's worth of outbound they do on a daily basis, I need an interface that would allow them to stick a comma delimited file or file(s) in every day via FTP, the file would contain call #'s, and some additional variables, and then the Asterisk box would schedule the calls. It would pull a voice file locally and deliver to answering machines or live call recipients. Looks like user interface is not a concern - if they are thinking of FTP text files. In this case, a simple script to kick off some call files should suffice. Won't take a week. (Search for call file.) But having to deal with answering machines is always tricky for any automation. Yuan Liu Cory Andrews -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lee Jenkins Sent: Tuesday, March 20, 2007 3:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk Automated Outbound Messaging Cory Andrews wrote: I have a client application looking for an Asterisk based solution. Client wants to deliver pre-recorded messages for a variety of clients. Wondering if anyone is offering an middleware for Asterisk for management of outbound messaging? Someone can correct me if I'm wrong, but I think a friend of mine mentioned that TrixBox has a gab cast function. It also shouldn't be that difficult to put together a script to do this. I actually have plans to do this myself, but no need for it just yet... If they want a decent interface to see the next caller before calling, you might want to have a database that reads in all the numbers, then users that grab the next non-checked number from the database. This also gives you the option of leaving notes with that call (such as calling back, etc). Then when ready, press the call button which creates a call file. Rob ___ --Bandwidth 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] error, install freePbx
On Tue, Mar 20, 2007 at 12:07:30PM +, Carlos Jerónimo wrote: Hi, i try install FreePbx by tuturial in http://www.voip-info.org/wiki/view/Instalaci%C3%B3n+de+Asterisk+en+Ubuntu+Server+Paso+a+Pasoview_comment_id=13443 but i have this error when i try install freepbx: #pear install db No. If at all, then use a package. Something of the sort of: apt-get install php4-pear No releases available for package pear.php.net/db Cannot initialize 'db' , invalid or missing package files Package db is not valid install failed Why this error? help me, please. AFAIK that tutorial is a bit obsolete. We're looking for testers for deb packages of freePBX from pkg-voip's buildserver, though. http://buildserver.net/ -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth 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] Microsoft launches first PABX
mitcheloc wrote: Is that FUD really necessary? No. Everyone can see this will be a disaster without the FUD. -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Asterisk Automated Outbound Messaging
Looks like user interface is not a concern - if they are thinking of FTP text files. In this case, a simple script to kick off some call files should suffice. Won't take a week. (Search for call file.) But having to deal with answering machines is always tricky for any automation. Yuan Liu Don't forget the other 'fun' issues related to auto-dialing with .call files (or AMI originate): Detecting and handling fax machines Figuring out whether a 'failed' call is a no answer or an invalid phone number (Yes, this is a tricky one, especially when using PRI) Getting correct CDR info back into the host system, if this is a requirement Establishing the calls is the easy part. Figuring out exactly what transpires AFTER the calls are originated is the true challenge. -MC ___ --Bandwidth 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 Automated Outbound Messaging
Sorry for my bad english,
I've developed a web interface for one of our customers, that allow them
to create lists of telephone numbers, (even from excel files), and then
a couple of scripts, one of them running on the background:
Script 1: Reads the file containing the telephone numbers and then
creates .call files for each number, and then put all the .call files on
a temp directory. You can start this process at any time. Even from a
cron job.
Script 2: Every specific time checks the temp directory, and if it
founds .call files, then it will copy the .call files to the outgoing
asterisk directory, but only allowing, for example 5 .calls at the time.
That way the asterisk only will let go 5 calls at the time. This script
must be running on the background.
This is script 2:
#!/bin/bash
INTERVALO=10
tmp=/tmp/calls #directory containing the .calls files
limite_canales=5 #number of simultaneus calls
asterisk=/var/spool/asterisk/outgoing
logger=/var/log/llamadas.log
fecha=`date`
function Generar_llamada {
outgoing=`ls -1 $asterisk | wc -l`
if [ $outgoing -eq 0 ] ; then
echo `date` No existen llamadas en este momento..
podemos incluir $limite_canales llamadas!! $logger
ls -1 $tmp /tmp/lista_llamadas_totales
tail -n $limite_canales /tmp/lista_llamadas_totales
/tmp/lista_llamadas
cat /tmp/lista_llamadas | while read line
do
mv $tmp/${line} $asterisk
done
fi
if [ $outgoing -gt 0 ] [ $outgoing -lt
$limite_canales ] ; then
let extras=$limite_canales-$outgoing
echo `date` Existen $outgoing llamadas en curso..
podemos incluir $extras llamadas $logger
ls -1 $tmp /tmp/lista_llamadas_totales
tail -n $extras /tmp/lista_llamadas_totales
/tmp/lista_llamadas
cat /tmp/lista_llamadas | while read line
do
mv $tmp/${line} $asterisk
echo `date` Archivo $line fue movido a
$asterisk $logger
done
fi
if [ $outgoing -ge $limite_canales ] ; then
echo `date` No existen llamadas Disponibles en este
momento.. favor de intentar mas tarde!! $logger
fi
}
while : ; do
call_files=`ls -1 $tmp | wc -l`
if [ $call_files -gt 0 ] ; then
echo `date` Comenzando con el procedimiento de
generacion de llamadas... $logger
Generar_llamada
else
echo `date` No existen cambios en el directorio $tmp
$logger
echo `date` A dormir por $INTERVALO segundos $logger
fi
sleep $INTERVALO
done
Hope it helps!
--
Ing. Arturo Ochoa N
Network Administrator
Electrosystems,
Yuan LIU escribió:
From: Rob Schall [EMAIL PROTECTED]
Date: Tue, 20 Mar 2007 16:00:01 -0500
Cory Andrews wrote:
These folks have 6-8 T's worth of outbound they do on a daily basis, I
need an interface that would allow them to stick a comma delimited
file
or file(s) in every day via FTP, the file would contain call #'s, and
some additional variables, and then the Asterisk box would schedule
the
calls. It would pull a voice file locally and deliver to answering
machines or live call recipients.
Looks like user interface is not a concern - if they are thinking of
FTP text files. In this case, a simple script to kick off some call
files should suffice. Won't take a week. (Search for call file.) But
having to deal with answering machines is always tricky for any
automation.
Yuan Liu
Cory Andrews
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Lee
Jenkins
Sent: Tuesday, March 20, 2007 3:33 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk Automated Outbound Messaging
Cory Andrews wrote:
I have a client application looking for an Asterisk based solution.
Client wants to deliver pre-recorded messages for a variety of
clients.
Wondering if anyone is offering an middleware for Asterisk for
management of outbound messaging?
Someone can correct me if I'm wrong, but I think a friend of mine
mentioned that TrixBox has a gab cast function.
It also shouldn't be that difficult to put together a script to do
this.
I actually have plans to do this myself, but no need for it just
yet...
If they want a decent interface to see the next caller before calling,
you might want to have a database that reads in all the numbers, then
users that grab the next non-checked number from the database. This
also gives you the option of leaving notes with that call (such as
calling back, etc). Then when ready, press the call button which
creates a call file.
Rob
___
--Bandwidth and Colocation provided by Easynews.com --
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Re: [asterisk-users] Problem with ATT Maintenance protocol in PRI connection, no B+D channels available
I've never seen a PRI dchannel on a T1 on a timeslot other than the 24th. Are you sure that it's really on channel 23? Matthew Fredrickson On Mar 20, 2007, at 8:54 AM, Kanelbullar wrote: Thanks for your answer, Bruno. However, the configuration you provided is for an E1 connection and we are using a T1, having channel 23 as D channel. Bruno De Luca [EMAIL PROTECTED] escreveu:d-channel is in midle bchan=1-15,17-31 dchan=16 loadzone = it defaultzone = it Kanelbullar wrote:Hi guys, We are experiencing a problem with a T1 PRI connection. After trying a number of variations in the configuration files, the behavior is always the same: no B channels come up and the D channel doesn't appear to be working well. We can see there are ATT Maintenance messages being exchanged by asterisk and the provider, CONNECT and CONNECT ACKNOWLEDGE, but that doesn't appear to be enough to bring the D and B channels properly up. Are there any messages missing? When we attempt to make a call, we can see the Q.931 SETUP message being sent. But shortly after we are getting a LAPD DISC message, which ends up originating a Q.931 DISCONNECT message, terminating the call. What could be the problem here? • Could there be any configuration issue on our side? • Does libpri provide complete support to the ATT Maintenance protocol or could this connection be incompatible? Any help would be highly appreciated. Many thanks in advance, Paulo PS: Configuration files, messages and pri debug snippets follow zaptel.conf loadzone = us defaultzone=us #Span 1: TE2/0/1 T2XXP (PCI) Card 0 Span 1 PRI_T1 span=1,0,0,esf,b8zs,crc4 bchan=1-23 dchan=24 zapata.conf [channels] group = 0 usecallingpres = yes switchtype = national context = inbound signalling = pri_cpe usecallerid = yes channel = 1-23 messages -- Mar 19 15:32:23 NOTICE[3306] cdr.c: CDR logging disabled, data will be lost. Mar 19 15:32:23 WARNING[3306] pbx_ael.c: Unable to open '/etc/asterisk/extensions.ael': No such file or directory Mar 19 15:32:23 WARNING[3306] pbx.c: Requested contexts didn't get merged Mar 19 15:33:17 WARNING[3322] chan_zap.c: No D-channels available! Using Primary channel 24 as D-channel anyway! Mar 19 15:33:58 WARNING[3322] chan_zap.c: No D-channels available! Using Primary channel 24 as D-channel anyway! Mar 19 15:33:58 WARNING[3366] app_dial.c: Unable to forward voice [...] pri debug span -- [ 00 01 0a 0a 03 01 00 07 01 01 c0 18 01 ac ] Informational frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000 EA: 1 N(S): 005 0: 0 N(R): 005 P: 0 10 bytes of data -- Restarting T203 counter Stopping T_203 timer Starting T_200 timer Protocol Discriminator: ATT Maintenance (3) len=10 Call Ref: len= 1 (reference 0/0x0) (Originator) Message type: CONNECT (7) [01 01 c0] IE: Change Status (len = 3) [18 01 ac] Channel ID (len= 3) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 1 ChanSel: As indicated in following octets ] (...) [ 02 01 0a 0c 03 01 00 0f 01 01 c0 18 01 ac ] Informational frame: SAPI: 00 C/R: 1 EA: 0 TEI: 000 EA: 1 N(S): 005 0: 0 N(R): 006 P: 0 10 bytes of data -- ACKing all packets from 5 to (but not including) 6 -- Since there was nothing left, stopping T200 counter -- Stopping T203 counter since we got an ACK -- Nothing left, starting T203 counter Protocol Discriminator: ATT Maintenance (3) len=10 Call Ref: len= 1 (reference 0/0x0) (Originator) Message type: CONNECT ACKNOWLEDGE (15) [01 01 c0] IE: Change Status (len = 3) [18 01 ac] Channel ID (len= 3) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 1 ChanSel: As indicated in following octets ] (...) Protocol Discriminator: Q.931 (8) len=40 Call Ref: len= 2 (reference 2/0x2) (Originator) Message type: SETUP (5) [04 03 80 90 a2] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: u-Law (34) [18 03 a9 83 82] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 2 ] [1e 02 80 83] Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: User (0) Ext: 1 Progress Description: Calling equipment is non-ISDN. (3) ] [6c 06 21 80 37 31 30 30] Calling Number (len= 8) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation:
Re: [asterisk-users] Refund from SellVoip?
I got money back around 6 months ago . It was a via paypal claim and hey didn't reply till paypal's deadline so i got $30 back . On 17/03/07, Ira [EMAIL PROTECTED] wrote: At 02:32 PM 3/16/2007, you wrote: You were able to cancel service with Sellvoip? That's impressive, that Actually, it's Voxee I tried to cancel and failed. I still use SellVOIP and it mostly works but support is a problem. I'm starting to use using Telasip more though as they work and have a POP only 19ms from here, a big advantage. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Problem with ATT Maintenance protocol in PRIconnection, no B+D channels available
I've never seen a PRI dchannel on a T1 on a timeslot other than the 24th. Are you sure that it's really on channel 23? I think he meant channel 23 of channels 0~23, aka the 24th channel. -MC Matthew Fredrickson On Mar 20, 2007, at 8:54 AM, Kanelbullar wrote: Thanks for your answer, Bruno. However, the configuration you provided is for an E1 connection and we are using a T1, having channel 23 as D channel. Bruno De Luca [EMAIL PROTECTED] escreveu:d-channel is in midle bchan=1-15,17-31 dchan=16 loadzone = it defaultzone = it Kanelbullar wrote:Hi guys, We are experiencing a problem with a T1 PRI connection. After trying a number of variations in the configuration files, the behavior is always the same: no B channels come up and the D channel doesn't appear to be working well. We can see there are ATT Maintenance messages being exchanged by asterisk and the provider, CONNECT and CONNECT ACKNOWLEDGE, but that doesn't appear to be enough to bring the D and B channels properly up. Are there any messages missing? When we attempt to make a call, we can see the Q.931 SETUP message being sent. But shortly after we are getting a LAPD DISC message, which ends up originating a Q.931 DISCONNECT message, terminating the call. What could be the problem here? * Could there be any configuration issue on our side? * Does libpri provide complete support to the ATT Maintenance protocol or could this connection be incompatible? Any help would be highly appreciated. Many thanks in advance, Paulo PS: Configuration files, messages and pri debug snippets follow zaptel.conf loadzone = us defaultzone=us #Span 1: TE2/0/1 T2XXP (PCI) Card 0 Span 1 PRI_T1 span=1,0,0,esf,b8zs,crc4 bchan=1-23 dchan=24 zapata.conf [channels] group = 0 usecallingpres = yes switchtype = national context = inbound signalling = pri_cpe usecallerid = yes channel = 1-23 messages -- Mar 19 15:32:23 NOTICE[3306] cdr.c: CDR logging disabled, data will be lost. Mar 19 15:32:23 WARNING[3306] pbx_ael.c: Unable to open '/etc/asterisk/extensions.ael': No such file or directory Mar 19 15:32:23 WARNING[3306] pbx.c: Requested contexts didn't get merged Mar 19 15:33:17 WARNING[3322] chan_zap.c: No D-channels available! Using Primary channel 24 as D-channel anyway! Mar 19 15:33:58 WARNING[3322] chan_zap.c: No D-channels available! Using Primary channel 24 as D-channel anyway! Mar 19 15:33:58 WARNING[3366] app_dial.c: Unable to forward voice [...] pri debug span -- [ 00 01 0a 0a 03 01 00 07 01 01 c0 18 01 ac ] Informational frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000 EA: 1 N(S): 005 0: 0 N(R): 005 P: 0 10 bytes of data -- Restarting T203 counter Stopping T_203 timer Starting T_200 timer Protocol Discriminator: ATT Maintenance (3) len=10 Call Ref: len= 1 (reference 0/0x0) (Originator) Message type: CONNECT (7) [01 01 c0] IE: Change Status (len = 3) [18 01 ac] Channel ID (len= 3) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 1 ChanSel: As indicated in following octets ] (...) [ 02 01 0a 0c 03 01 00 0f 01 01 c0 18 01 ac ] Informational frame: SAPI: 00 C/R: 1 EA: 0 TEI: 000 EA: 1 N(S): 005 0: 0 N(R): 006 P: 0 10 bytes of data -- ACKing all packets from 5 to (but not including) 6 -- Since there was nothing left, stopping T200 counter -- Stopping T203 counter since we got an ACK -- Nothing left, starting T203 counter Protocol Discriminator: ATT Maintenance (3) len=10 Call Ref: len= 1 (reference 0/0x0) (Originator) Message type: CONNECT ACKNOWLEDGE (15) [01 01 c0] IE: Change Status (len = 3) [18 01 ac] Channel ID (len= 3) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 1 ChanSel: As indicated in following octets ] (...) Protocol Discriminator: Q.931 (8) len=40 Call Ref: len= 2 (reference 2/0x2) (Originator) Message type: SETUP (5) [04 03 80 90 a2] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: u-Law (34) [18 03 a9 83 82] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 2 ] [1e 02 80 83] Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: User (0)
Re: [asterisk-users] Microsoft launches first PABX
Next generation bot nets -- forget about spam, telemarketing is the next viral (literally) marketing concept! On 3/20/07, Jon Pounder [EMAIL PROTECTED] wrote: Quoting C F [EMAIL PROTECTED]: Awesome, the first PABX virus is just around the corner now that M$ has some bait for it to infect. In a world without borders we don't need windows or gates. ___ --Bandwidth 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] SIP/Polycom Issue, Asterisk 1.2.16, calls dropped
I've been running the 8/1/2004 Head release up until a little over a week ago. I was forced to due to a card failure to upgrade to 1.2.16 without any advance preparation or testing (most of my connections are via satellite to all corners of the globe with high latency). Up until the upgrade I was running with very few issues. Since the upgrade I have been experiencing strange issues with my Polycom SP-601 phones. My customers attempt to get their voicemail and Asterisk drops their connection ~15 seconds after they dial VM. I have captured a SIP debug and included it (somewhat sanitized). I'm not a SIP guru, but I can see the 15 second timer being set and I see repeated INVITEs being sent without any acks. OPTIONs are being sent and acked. The remote SIP phone is 'eden-1000a' and the voicemail extension is 9990. *This worked just fine up until the upgrade.* Does this ring a bell with anyone out there??? Tim McKee tmckee at sdnglobal dot com SDN Global == pbx*CLI sip debug peer eden-1000a SIP Debugging Enabled for IP: 10.253.4.50:5060 pbx*CLI -- SIP read from 10.253.4.50:5060: INVITE sip:[EMAIL PROTECTED];user=phone SIP/2.0 Via: SIP/2.0/UDP 10.253.4.50;branch=z9hG4bK8ed5192B7E6AF From: eden-1000a sip:[EMAIL PROTECTED];tag=D4964260-95FB99E3 To: sip:[EMAIL PROTECTED];user=phone CSeq: 1 INVITE Call-ID: [EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_601-UA/2.0.3.0127 Supported: 100rel,replaces Allow-Events: talk,hold,conference Max-Forwards: 70 Content-Type: application/sdp Content-Length: 245 v=0 o=- 978307756 978307756 IN IP4 10.253.4.50 s=Polycom IP Phone c=IN IP4 10.253.4.50 t=0 0 m=audio 2228 RTP/AVP 0 18 8 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 --- (14 headers 11 lines) --- Using INVITE request as basis request - [EMAIL PROTECTED] Sending to 10.253.4.50 : 5060 (NAT) Reliably Transmitting (no NAT) to 10.253.4.50:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.253.4.50;branch=z9hG4bK8ed5192B7E6AF;received=10.253.4.50 From: eden-1000a sip:[EMAIL PROTECTED];tag=D4964260-95FB99E3 To: sip:[EMAIL PROTECTED];user=phone;tag=as7f808f0f Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Proxy-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=2584558d Content-Length: 0 --- Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms Found user 'eden-1000a' pbx*CLI -- SIP read from 10.253.4.50:5060: INVITE sip:[EMAIL PROTECTED];user=phone SIP/2.0 Via: SIP/2.0/UDP 10.253.4.50;branch=z9hG4bK8ed5192B7E6AF From: eden-1000a sip:[EMAIL PROTECTED];tag=D4964260-95FB99E3 To: sip:[EMAIL PROTECTED];user=phone CSeq: 1 INVITE Call-ID: [EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_601-UA/2.0.3.0127 Supported: 100rel,replaces Allow-Events: talk,hold,conference Max-Forwards: 70 Content-Type: application/sdp Content-Length: 245 v=0 o=- 978307756 978307756 IN IP4 10.253.4.50 s=Polycom IP Phone c=IN IP4 10.253.4.50 t=0 0 m=audio 2228 RTP/AVP 0 18 8 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 --- (14 headers 11 lines) --- Ignoring this INVITE request pbx*CLI -- SIP read from 10.253.4.50:5060: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 10.253.4.50;branch=z9hG4bK8ed5192B7E6AF From: eden-1000a sip:[EMAIL PROTECTED];tag=D4964260-95FB99E3 To: sip:[EMAIL PROTECTED];user=phone;tag=as7f808f0f CSeq: 1 ACK Call-ID: [EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_601-UA/2.0.3.0127 Max-Forwards: 70 Content-Length: 0 --- (11 headers 0 lines) --- pbx*CLI -- SIP read from 10.253.4.50:5060: INVITE sip:[EMAIL PROTECTED];user=phone SIP/2.0 Via: SIP/2.0/UDP 10.253.4.50;branch=z9hG4bK82926abd205366FA From: eden-1000a sip:[EMAIL PROTECTED];tag=D4964260-95FB99E3 To: sip:[EMAIL PROTECTED];user=phone CSeq: 2 INVITE Call-ID: [EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_601-UA/2.0.3.0127 Supported: 100rel,replaces Allow-Events: talk,hold,conference Proxy-Authorization: Digest username=eden-1000a, realm=asterisk, nonce=2584558d, uri=sip:[EMAIL PROTECTED];user=phone, response=d9b3ca0769228d580b8877300d1e4ef3, algorithm=MD5 Max-Forwards: 70 Content-Type: application/sdp Content-Length: 245 v=0 o=- 978307756 978307756 IN IP4 10.253.4.50
Re: [asterisk-users] SIP/Polycom Issue, Asterisk 1.2.16, calls dropped
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Are you using Answer() before VoiceMailMain()? Stu Timothy McKee wrote: I've been running the 8/1/2004 Head release up until a little over a week ago. I was forced to due to a card failure to upgrade to 1.2.16 without any advance preparation or testing (most of my connections are via satellite to all corners of the globe with high latency). Up until the upgrade I was running with very few issues. Since the upgrade I have been experiencing strange issues with my Polycom SP-601 phones. My customers attempt to get their voicemail and Asterisk drops their connection ~15 seconds after they dial VM. I have captured a SIP debug and included it (somewhat sanitized). I'm not a SIP guru, but I can see the 15 second timer being set and I see repeated INVITEs being sent without any acks. OPTIONs are being sent and acked. The remote SIP phone is 'eden-1000a' and the voicemail extension is 9990. *This worked just fine up until the upgrade.* Does this ring a bell with anyone out there??? Tim McKee tmckee at sdnglobal dot com SDN Global == pbx*CLI sip debug peer eden-1000a SIP Debugging Enabled for IP: 10.253.4.50:5060 pbx*CLI -- SIP read from 10.253.4.50:5060: INVITE sip:[EMAIL PROTECTED];user=phone SIP/2.0 Via: SIP/2.0/UDP 10.253.4.50;branch=z9hG4bK8ed5192B7E6AF From: eden-1000a sip:[EMAIL PROTECTED];tag=D4964260-95FB99E3 To: sip:[EMAIL PROTECTED];user=phone CSeq: 1 INVITE Call-ID: [EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_601-UA/2.0.3.0127 Supported: 100rel,replaces Allow-Events: talk,hold,conference Max-Forwards: 70 Content-Type: application/sdp Content-Length: 245 v=0 o=- 978307756 978307756 IN IP4 10.253.4.50 s=Polycom IP Phone c=IN IP4 10.253.4.50 t=0 0 m=audio 2228 RTP/AVP 0 18 8 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 --- (14 headers 11 lines) --- Using INVITE request as basis request - [EMAIL PROTECTED] Sending to 10.253.4.50 : 5060 (NAT) Reliably Transmitting (no NAT) to 10.253.4.50:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.253.4.50;branch=z9hG4bK8ed5192B7E6AF;received=10.253.4.50 From: eden-1000a sip:[EMAIL PROTECTED];tag=D4964260-95FB99E3 To: sip:[EMAIL PROTECTED];user=phone;tag=as7f808f0f Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Proxy-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=2584558d Content-Length: 0 --- Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms Found user 'eden-1000a' pbx*CLI -- SIP read from 10.253.4.50:5060: INVITE sip:[EMAIL PROTECTED];user=phone SIP/2.0 Via: SIP/2.0/UDP 10.253.4.50;branch=z9hG4bK8ed5192B7E6AF From: eden-1000a sip:[EMAIL PROTECTED];tag=D4964260-95FB99E3 To: sip:[EMAIL PROTECTED];user=phone CSeq: 1 INVITE Call-ID: [EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_601-UA/2.0.3.0127 Supported: 100rel,replaces Allow-Events: talk,hold,conference Max-Forwards: 70 Content-Type: application/sdp Content-Length: 245 v=0 o=- 978307756 978307756 IN IP4 10.253.4.50 s=Polycom IP Phone c=IN IP4 10.253.4.50 t=0 0 m=audio 2228 RTP/AVP 0 18 8 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 --- (14 headers 11 lines) --- Ignoring this INVITE request pbx*CLI -- SIP read from 10.253.4.50:5060: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 10.253.4.50;branch=z9hG4bK8ed5192B7E6AF From: eden-1000a sip:[EMAIL PROTECTED];tag=D4964260-95FB99E3 To: sip:[EMAIL PROTECTED];user=phone;tag=as7f808f0f CSeq: 1 ACK Call-ID: [EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_601-UA/2.0.3.0127 Max-Forwards: 70 Content-Length: 0 --- (11 headers 0 lines) --- pbx*CLI -- SIP read from 10.253.4.50:5060: INVITE sip:[EMAIL PROTECTED];user=phone SIP/2.0 Via: SIP/2.0/UDP 10.253.4.50;branch=z9hG4bK82926abd205366FA From: eden-1000a sip:[EMAIL PROTECTED];tag=D4964260-95FB99E3 To: sip:[EMAIL PROTECTED];user=phone CSeq: 2 INVITE Call-ID: [EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_601-UA/2.0.3.0127 Supported: 100rel,replaces Allow-Events: talk,hold,conference Proxy-Authorization: Digest username=eden-1000a, realm=asterisk, nonce=2584558d,
[asterisk-users] wrong values in duration and billsec in CDR
Hi to all, I was looking in google and also in this mailing list, but I dont find the solution to my problem, so I subscribe me to the list in order to post this e-mail and find the solution. This is the scenario: GSM Phone - GSM Network TDM2406E --- ASterisk 1.4.0 (*) VoIP Provider --- Sip Phone or H323 Phone The problem is that I am generating calls from SIP and also h323 (using ooh323), and always I saw differences between duration time and billsecs just for 4 or 3 seconds. Altought the difference is much more, I mean I just call from SIP and H323 clients and always I saw the same behaviour. When I generate the call I wait to pickup the cell phone almost 10 secs, the right time should be something like 30 secs, but I saw duration = 50, and billsec = 47. This is a very weird behaviour and I was trying to modify zaptel.conf but I can't find what my problem is. I guess the problem is that this time is being counted just from Voip domain, and not into the Zaptel domain. Maybe some of you could guide me to solve this problem. Any help will be so much appreciated Rgds, Jovanny Saravia [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel silly issue
On 3/19/07, Brad Sumrall [EMAIL PROTECTED] wrote: ...music on hold... Brad Music on hold support is present, you can also add MP3 support with asterisk-addons package, are you using MP3 without the correct format installed? www.asterisk.org and download the add-ons package, read the docs while you are at it for a problem-free installation. ___ --Bandwidth 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] Teliax problems, they say use SIP, more mature better working than IAX
I really have lost loads of faith for IAX. No authority found and Rejected connect attempt messages for no apparent reason. Sometimes computability issues between asterisk versions. Fax/T.38 support?? But I have no complaints about when it actually does work. Not that Asterisk has the best SIP implementation either... ___ --Bandwidth 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] codec_zap and Asterisk 1.4.1
Jeremiah Millay wrote: I'm a little confused. I'm running a sangoma a200 card in my server. I thought I needed codec_zap. Do I need to wait until zaptel-1.4.1 gets released to be able to compile this or do I just not need it all together? Any insight would be appreciated. You don't need codec_zap unless you are using a TC400B transcoder card. You need chan_zap if you are using Zaptel for channels. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
