Re: [Asterisk-Users] CallerID rewrite php AGI Script
Just looked at my gmail archive of this mailing list. Just happened to remember a thread about this: On 5/8/05, Jay Milk [EMAIL PROTECTED] wrote: Mine does business lookups properly, and also uses a mySQL database to cache results (and allows you to store your own results): http://www.muware.com/asterisk/ And yes, it does both google and 411.com, and falls back to telcodata.us to get CO information if the other two come back empty. Since it's already on there, it stores the address in the DB as well, if google or 411 return it. On 7/14/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hello all.I am looking for the great callerID rewrite script that does the 411lookup and then stores the information in a database.If there is information in the Database for the callerid coming in, then use that and pass it along to the phone.I lost my entire system hard drive this week, and slowly rebuilding. Thisscript wasn't in the most recent backup :( :( :(Please help :)thanks. Ben___Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] NO calling tone
Thank you Michiel.
I tried to remove m and use r , but still not working, after I change r
to R , it is working. Anybody know why?
Michiel van Baak wrote:
On 11:12, Wed 13 Jul 05, Bill Wong wrote:
Can you show me the example, i am newbie.NOt sure whether the code i
modified is correct or not..
my code as below..
exten = 671042,1,Dial(${PHONES1},20,Ttmr)
loose the m.
m = provide music while ringing
r = provide ring sound while ringing.
Using both is conflicting and will result in silence while
ringing.
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[Asterisk-Users] VOIP phone, how to use with asterisk ??
Hi, I tested asterisk server with Xpro program, and all the function working well ( like 3 way calling, transfer ). But on the VOIP phone, I don't know press which key for 3 way calling function and transfer function... Can anybody teach me ? thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Is soekris good?
Hi, I have a soekris 4801 since some days. I use it with a FritzCard-USB and an internal HFC-Card (NT Mode). Everything is working, but I still havn't had time for performance test. Only thing I tested, was two ISDN channels via FritzCard in a conference room. CPU usage was nearly 70% I hope next weekend I'll find more time. What WiFi phone do you want to use? I tried a ZyXEL P2000W, but voice quality was very bad. regards Jens We found at the wiki a link to soekris and wonder if it is good? Is anybody using it and can share some experience, please? We would like to use it as a small PBX including a wireless access point, so that we can also use WiFi phones. bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SpanDSP+astfax with multiple fax pages
We have still problems also when we use this command to convert the document to the right tiff format. I sended 748 faxes, I think 213 where ok, 358 got only the first page, 177 have failed (no answer busy or something like that) You can say that this is a very bad result! We also can only send 5 faxes at the same time but we have 30 lines for sending. (else we get: Call failed to go through, reason 0) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RE: AgentCallbackLogin Question
thanks, this solves not only the agent id problem, i could also add extension roaming too. if i set the updatecdr value to yes, which column in the cdr table is it recorded? On 7/13/05, Jason Kawakami [EMAIL PROTECTED] wrote: I'm looking for a way to capture the Agent ID after login, to keep track which agent is associated in a certain call. --check out updatecdr=yes in agents.conf Jason Kawakami www.optellabs.com Salt Lake City, UT ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Festival questions
Has anyone had any luck in changing the voices for Festival and Asterisk? I have Festival installed and working, but can not get the voice different from the default. Thanks, Jason -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jim Archer Sent: Wednesday, July 13, 2005 10:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Festival questions I'm working on this now. I don't expect it to be too useful though. --On Wednesday, July 13, 2005 3:47 PM -0400 [EMAIL PROTECTED] wrote: Hi, Is it possible to setup an Asterisk system that can allow someone to dial in using a DID and listen to their e-mail? Has anyone done this? Thanks, Mike C. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Faxing Suggestions
If you have an solutions that have worked well for you in the past, I would love to hear them. On one of my customer sites, I had exactly the same problem. One PRI to the Telco, local SIP phones and a 4 port FXS TDM400P card. The faxes were totally unreliable and I couldn't do anything to fix it. After that I did a lot of experimenting and found a solution. I used an ISDN BRI card (Sirrix 4 port BRI, Junghanns QuadBRI or OctoBRI work fine although I vote for a Sirrix one) and I hooked up an ISDN S0 to analog converter to it (DeTeWe TA33clip works fine). It had a few glitches, but with a newer versions bristuff and/or newer Sirrix card drivers everything works like a charm. Ivan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_sccp new release
Andre Normandin ha scritto: Is there a place I can go that documents all the options in the sccp.conf file? I'm writing the new config parser, so sccp.conf structure will soon change. I'll write the documentations Sergio ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] DBput from the web?
Does anybody has a php code for using DBput (DBget, DBdel) from a web interface, which database is used for astrisk? I don't have anything similar, but instead of using * internal DB maybe you should consider using MySQL. Ivan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SpanDSP rxfax, no tiff.
I may be a little late on this, but what permissions are on
/usr/local/sbin/mailfax?
I have a similar set up to execute a mysql query to grab
the email address based on DNIS (PRI T1 with multiple numbers on one circuit)
and then email the fax to the destination. I set the perm to 755 on the script
so everyone/thing can execute.
Also, what are the perms on
/var/spool/asterisk/asterisk-fax?
Can you run the script from the command line by passing it
the appropriate values (i.e. /usr/local/sbin/mailfax
/var/spool/asterisk/asterisk-fax/#.#.tiff [EMAIL PROTECTED]?
In the event that something weird is going on with the
command line parameters, here are some considerations:
If the folder for the TIFFs is always the same, you could do a Set (or
SetVar depending on your Asterisk build) to have the UNIQUEID passed only to
the script
If the email reciepients are always on the same domain, you need to
only pass the name portion of the email address
For example:
Extensions.conf
section ---
[fax]
exten =
s,1,Answer
exten =
s,2,Macro(faxreceive)
;exten =
h,1,system(/usr/local/sbin/mailfax ${FAXFILE} ${EMAILADDR})
This line could go away
[macro-faxreceive]
exten =
s,1,Set(FAX_OUT=${UNIQUEID})
exten = s,2,Set(FAXFILE=/var/spool/asterisk/asterisk-fax/${FAX_OUT}.tif)
exten = s,3,rxfax(${FAXFILE})
exten = s,4,system(/usr/local/sbin/mailfax
${FOX_OUT} MyName)
;exten =
s,3,Set([EMAIL PROTECTED])
This line could go away
Assuming #!/bin/bash
;)
#!/bin/bash
# $ARG1 is the
TIFF file name
# $ARG2 is the
name of the domain email user
EMAIL_ADDR=$2"@mycompany.com"
FAX_FILE="/var/spool/asterisk/asterisk-fax/"$1
# Do the sendmail
thing here
#--
Just my 40
cents.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rob
DanzSent: Wednesday, July 13, 2005 8:18 AMTo:
asterisk-users@lists.digium.comSubject: [Asterisk-Users] SpanDSP
rxfax, no tiff.
Hello,
Let me start by saying I have
checked the wiki and the archives and did find some relative information.
I tried the suggestions in those threads, but still have the same
problem.
Im using the CVS Asterisk from July
11, 2005.
Redhat
FC2
SpanDSP
0.0.2pre18
Libtiff
3.5.7
Digium PCI card 1 FXO,
1FXS.
I have a single POTS line coming,
but I have 2 numbers and am using distinctive ring detection in
*.
When you call my fax number, the
ring detection does work, and does send it to the fax context
correctly.
The debugs show the call is
answered, rxfax is invoked and it is trying to write to the fax file.
After the sending party hangs up, it tries to execute a script that will
ultimately mail me the fax file. But since the tiff file isnt there to
begin with, that fails. The permissions on that folder are 777 for now so
permissions arent the problem.
I saw a post by Steve Underwood from
last year on a similar problem, but it was looking like timing slips on the
T1/E1 for that user Im just using a POTS line though. Ive also done
ztmonitor to look at the Rx and Tx levels. Rx is a little hotter than Tx,
but theyre both well on the right hand side of the scale.
Any help is appreciated.
Debugs extensions.conf excerpt are below.
Thanks,
Rob
Debug output
---
Jul 13 10:04:34 NOTICE[7975]:
chan_zap.c:5759 ss_thread: Got event 2
(Ring/Answered)...
-- Detected ring
pattern: 93,0,0
-- Distinctive
Ring matched context fax
-- Executing
Answer("Zap/4-1", "") in new stack
-- Executing
Macro("Zap/4-1", "faxreceive") in new stack
-- Executing
Set("Zap/4-1", "FAXFILE=/var/spool/asterisk/asterisk-fax/1121267067.12.tif") in
new stack
-- Executing
RxFAX("Zap/4-1", "/var/spool/asterisk/asterisk-fax/1121267067.12.tif") in new
stack
-- Executing
System("Zap/4-1", "/usr/local/sbin/mailfax
/var/spool/asterisk/asterisk-fax/1121267067.12.tif ") in new
stack
Jul 13 10:05:03 WARNING[7975]:
app_system.c:75 system_exec_helper: Unable to execute '/usr/local/sbin/mailfax
/var/spool/asterisk/asterisk-fax/1121267067.12.tif
'
-- Hungup
'Zap/4-1'
Extensions.conf section
---
[fax]
exten =
s,1,Answer
exten =
s,2,Macro(faxreceive)
exten =
h,1,system(/usr/local/sbin/mailfax ${FAXFILE}
${EMAILADDR})
[macro-faxreceive]
exten =
s,1,Set(FAXFILE=/var/spool/asterisk/asterisk-fax/${UNIQUEID}.tif)
exten =
s,2,rxfax(${FAXFILE})
exten =
s,3,Set([EMAIL PROTECTED])
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[Asterisk-Users] Panasonic PBX -to- Sirrix BRI: Numbers getting echoed/duplicated
Hi guys, How's things going ? Got a bit of a weird one here that I've been unable to solve. I have a Panasonic PBX linked to a Sirrix Quad BRI card that is running in TE (ptp) mode in a Asterisk box -this then links through Internet to another Asterisk box via IAX2. When a user on the Panasonic PBX system dials the extension of my Sirrix Asterisk box, Asterisk answers and says "Please dial the number of the person you are looking for". This is done with cmd "Background". When this user enters an extension number to call the numbers that get picked up by Asterisk are repeated/echoed. For example, if a user enters "19" at the voice prompt, Asterisk picks it up as "1199" and tries to then dial "1199" out to the remote Asterisk server. Any ideas what causes this ? Kindest regardsDavid Wilson___D c D a t aTel +27 33 342 7003Fax +27 33 345 4155Cell +27 82 4147413http://www.dcdata.co.za[EMAIL PROTECTED]Powered by Linux, driven by passion ! ___ "Computers are not intelligent. They only think they are." ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Multiple NICs on Asterisk box
Kevin P. Fleming wrote: Zoltan Szecsei wrote: I've traced the problem to be with the firewall and the fact that I have 2 NICs in the box. Now that I have opened port 4569 on both interfaces, asterisk seems happy *but* does anyone know how to force SuSE 9.3 to always bring up a specific NIC before the otherone? SUSE Pro 9.3 assigns IP addresses to NICs based on their MAC addresses, not their slot positions or load order (unless you've done something funky). You should not have any problem with the wrong NIC getting a particular IP address, and Asterisk does not care at all which one is 'eth0'. I wish this were true, but, believe me, on reboot, sometime the pci 3com card gets eth0 and sometime the onboard. Who says computers re-iterate perfectly :-) ? BTW: I need the pci card to be eth0 (sensed first). Zoltan -- == Geograph (Pty) Ltd P.O. Box 31255 Tokai 7966 Tel:+27-21-7018492 Fax:+27-86-6115323 Mobile: +27-83-6004028 == ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Multiple NICs on Asterisk box
Ken Godee wrote: On reboot sometimes my onboard gigabit nic gets eth0 and sometimes the pci 3COM gets eth0 and this causes havoc with another piece of SW I run. Is it actually ethx getting flipped or the ip addresses? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ethx gets flipped -- == Geograph (Pty) Ltd P.O. Box 31255 Tokai 7966 Tel:+27-21-7018492 Fax:+27-86-6115323 Mobile: +27-83-6004028 == ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CVS HEAD voicemailbox full error
Anyone else has problems with CVS HEAD's from today with voicemail hanging up silently without any debug/error messages when checked? It also keeps insisting that the user's voice mailbox is full and can't store more messages even if I clear/rebuild the /var/spool/asterisk/voicemail stuff. I've tried falling back to voicemail.conf entries from realtime voicemail with the same result. Thanks, Vahan begin:vcard fn:Vahan Yerkanian n:Yerkanian;Vahan org:ARMINCO Global Telecommunications;Head, Research Development dept. adr:;;28, Isahakian ave., PO BOX 10;Yerevan;;375009;Armenia email;internet:[EMAIL PROTECTED] x-mozilla-html:FALSE url:http://www.arminco.com/ version:2.1 end:vcard ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Multiple NICs on Asterisk box
On Thu, 2005-07-14 at 09:01 +0200, Zoltan Szecsei wrote: Ken Godee wrote: On reboot sometimes my onboard gigabit nic gets eth0 and sometimes the pci 3COM gets eth0 and this causes havoc with another piece of SW I run. I seem to remember having this type of thing before, only I had two identical NICs, because they were good ones :) Have you tried creating an alias in modprobe.conf (2.6) or modules.conf (2.4) alias eth0 b44 alias eth1 e100 in my case -- Dave Cotton [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to integrate the Call Pickup with CID info feature in the release tree of Asterisk?
Hi, we really need the feature Call Pickup with CID info http://www.voip-info.org/wiki-Asterisk+bounty+Call+Pickup+with+CID+info.+-+SIP in the current Asterisk release because we have a newer TE405P card which needs 1.0.8 or newer to work. The call pickup patch only works for 1.0.7. Who is responsible for such a wish? Regards, Kib ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CAPI in PTP mode not answering, dial out fine
Hi list,
I am using Asterisk in a small systems with an AVM C4 card, we first had
one ISDN line, (ptmp), which we upgraded to
2 ISDN with 1 number (so no DID's) This runs in ptp mode.
Calling out works fine on all 4 channels, but when I call in, I get
*CLI Jul 13 09:44:59 ERROR[13635]: chan_capi.c:1695 pipe_msg: did not
find device for msn = 299450707
(my number is 0299-450707
The call gets to the C4 card, my kernel logs:
isdn_net: call from 62411 - 0 299450707 ignored
isdn_tty: call from 62411 - 299450707 ignored
capidrv-1: incoming call 62411,1,0,299450707 ignored
my capi.conf:
[interfaces]
isdnmode=ptp
mode=immediate
msn=299450707
incomingmsn=299450707
controller=1,2
softdtmf=1
context=outbound
echosquelch=1
echocancel=yes
echotail=64
callgroup=1
devices=4
extensions.conf (part)
[outbound]
ignorepat = 0
exten = _0.,1,Ringing
exten = _0.,2,Dial(CAPI/299450707:${EXTEN:1})
exten = _0.,3,Congestion
[default]
exten = s,1,Dial(sip/20,25)
exten = s,2,Dial(sip/21,25)
exten = _299450707,1,Goto(s,1)
exten = 0299450707,1,Goto(s,1)
exten =_450707,1,Goto(s,1)
exten = 299450707,1,Goto(s,1)
include = outbound
As you can see I've tried every possible option to get asterisk to match
the MSN,
but the because the error says no _DEVICE_ found, I don;t think it will
even make it to the extensions.conf.
I use asterisk 1.07 with chan_capi 0.35
Kind regards,
Joop Marijne
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Re: [Asterisk-Users] NoOp
And is there some bit of information I get at verbose level 255 that I don't get at 254? It just seems like a lot of levels. MARK. John Novack wrote: MF Hulber wrote: It's a little odd. Something like asterisk -v4 seems more appropriate. You can also use set verbose level so that you don't have to restart your console session to change the verbosity. I really don't know what the maximum effective verbose level is. MARK. 255 JN George Garvey wrote: On Sun, Jul 10, 2005 at 09:49:37PM -0400, MF Hulber wrote: Maybe it shows up after a certain verbosity level. Try asterisk -r When I do that NoOps always show up. Looks like you're right. Guess I never used enough v's ;) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_sccp new release
Hi,
I have a lot of compile problems with your version, because I have to use
gcc-2.95.
The problem is, that you should decleare variables first.
gcc-3.x accept something like this:
void sccp_channel_set_calledparty(sccp_channel_t * c, char *name, char
*number)
{
if (!c)
return;
sccp_device_t * d = c-device;
but gcc-2.95 doesn't
Correct syntax would be:
void sccp_channel_set_calledparty(sccp_channel_t * c, char *name, char
*number)
{
sccp_device_t * d;
if (!c)
return;
d = c-device;
I can send you my diffs, if you like.
regards
Jens
Quoting Sergio Chersovani [EMAIL PROTECTED]:
http://chan-sccp.berlios.de/
20050713 ftp://ftp.berlios.de/pub/chan-sccp/chan_sccp-20050713.tar.bz2
I didn't have a spare 7960 to use this week, so maybe some line issue is
still present.
- fixed a memory leak on database updates (dnd, cfwd*)
- fixed old memory leak on unload (now unload chan_sccp.so and load
chan_sccp.so work. It does reload the config when asterisk is running)
- socket stuff has been totally rewritten
- added sccp show sessions (cli command)
- modified the output of sccp show channels (use it to understand what
chan_sccp is doing with channels)
- rewrite of asterisk codecs 2 skinny translation
- modified the calls hangup system (more stable)
- minor changes on the native transfer (now the call on a failed
transfer status will ring back when you put onhook the phone - useful
for no display phones)
- many minor changes
How to build:
wget ftp://ftp.berlios.de/pub/chan-sccp/chan_sccp-20050713.tar.bz2
tar xvjf chan_sccp-20050713.tar.bz2
cd chan_sccp-20050713
make clean; make install
modules.conf
load = chan_sccp.so
noload = chan_skinny.so
edit sccp.conf
if you have compile errors try this:
rm /usr/include/asterisk/*
cd asterisk
make upgrade
cd chan_sccp-20050713
make clean; make install
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Re: [Asterisk-Users] DBput from the web?
Ivan Meic (Vox Mundi) wrote: Does anybody has a php code for using DBput (DBget, DBdel) from a web interface, which database is used for astrisk? I don't have anything similar, but instead of using * internal DB maybe you should consider using MySQL. Do you have an example how to use MySQL in Asterisk? bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CAPI in PTP mode not answering, dial out fine
On Thu, 14 Jul 2005, asterisk wrote:
Hi list,
I am using Asterisk in a small systems with an AVM C4 card, we first had one
ISDN line, (ptmp), which we upgraded to
2 ISDN with 1 number (so no DID's) This runs in ptp mode.
Calling out works fine on all 4 channels, but when I call in, I get
*CLI Jul 13 09:44:59 ERROR[13635]: chan_capi.c:1695 pipe_msg: did not find
device for msn = 299450707
...
my capi.conf:
[interfaces]
isdnmode=ptp
mode=immediate
msn=299450707
incomingmsn=299450707
controller=1,2
softdtmf=1
context=outbound
Your context is 'outbound', but
echosquelch=1
echocancel=yes
echotail=64
callgroup=1
devices=4
extensions.conf (part)
[outbound]
ignorepat = 0
exten = _0.,1,Ringing
exten = _0.,2,Dial(CAPI/299450707:${EXTEN:1})
exten = _0.,3,Congestion
here in 'outbound' there is no match to your msn.
As you can see I've tried every possible option to get asterisk to match the
MSN,
but the because the error says no _DEVICE_ found, I don;t think it will even
make it to the extensions.conf.
This message is confusing here, but it seems that the match is not found in
extensions.conf.
I use asterisk 1.07 with chan_capi 0.35
Maybe you want to try chan_capi-cm on sourceforge...
Armin
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Re: [Asterisk-Users] chan_sccp new release
asterisk_on_oelf ha scritto: I have a lot of compile problems with your version, because I have to use gcc-2.95. Yes, it's a performance choice to support only gcc version = 3.0 I will put a compiled chan_sccp module on my page for older gcc compilers Sergio ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Panasonic PBX -to- Sirrix BRI: Numbers getting echoed/duplicated
On Thu, 14 Jul 2005, David Wilson wrote: I have a Panasonic PBX linked to a Sirrix Quad BRI card that is running in TE (ptp) mode in a Asterisk box - this then links through Internet to another Asterisk box via IAX2. When a user on the Panasonic PBX system dials the extension of my Sirrix Asterisk box, Asterisk answers and says Please dial the number of the person you are looking for. This is done with cmd Background. When this user enters an extension number to call the numbers that get picked up by Asterisk are repeated/echoed. For example, if a user enters 19 at the voice prompt, Asterisk picks it up as 1199 and tries to then dial 1199 out to the remote Asterisk server. One possible cause is that Asterisk receives the digits both as isdn indications (out of band) and as dtmf. Are you sure you have answered the line? On a bri link audio can be passed even without the line being answered. Before the line is answered Asterisk can receive overlap digits. While in overlap reception mode in band (dtmf) digits are ignored. Yuo may want to enable overlap digits nn the link to the Panasonic. I am not familiar with this particular BRI card. If it is not based on zaptel then the configuration will have to be made elsewhere. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] *** install error
i've installed asterisk yesterday for the first time. i did make and make install for all the directory and after MAKE SAMPLES but when asterisk start i receive message: [app_readfile.so]Jul 14 10:21:17 WARNING[4069]: loader.c:258 ast_load_resource: /usr/lib/asterisk/modules/app_readfile.so: undefined symbol: ast_register_file_version Jul 14 10:21:17 WARNING[4069]: loader.c:440 load_modules: Loading module app_readfile.so failed! how can i resolve it? thanks luca _ Ricerche online più semplici e veloci con MSN Toolbar! http://toolbar.msn.it/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_sccp new release
I have a lot of compile problems with your version, because I have to use gcc-2.95. Yes, it's a performance choice to support only gcc version = 3.0 I will put a compiled chan_sccp module on my page for older gcc compilers Maybe you could use my attached diffs for further development. It works for 2.95 and I hope for 3.x too. chan_sccp-2.95-diff Description: Binary data ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Multiple NICs on Asterisk box
James Oakley wrote: On Wednesday 13 July 2005 12:40 pm, Zoltan Szecsei wrote: Hi All, Long time no chat ;-) Asterisk 1.0.9 (sometimes) won't authenticate IAX phones after re-boot of SuSE 9.3 box I've traced the problem to be with the firewall and the fact that I have 2 NICs in the box. Now that I have opened port 4569 on both interfaces, asterisk seems happy *but* does anyone know how to force SuSE 9.3 to always bring up a specific NIC before the otherone? On reboot sometimes my onboard gigabit nic gets eth0 and sometimes the pci 3COM gets eth0 and this causes havoc with another piece of SW I run. Look at the ifcfg-* files under /etc/sysconfig/network. There should be one for each of your interfaces. Add a persistent name to each of them like so: PERSISTENT_NAME='external' Call them whatever you want. Good choices are 'external', 'internal', etc. Now instead of eth0 and eth1 you have more logical names that will always refer to the same interfaces no matter which order they came up in. Hope that helps, Thanks - I did notice this as I searched the archives before posting my question, but this wont help my other piece of SW that authenticates against the MAC address of the 1st NIC sensed (see my last paragraph) :-( (now if macchager could be made to work reliably under SuSE 9.3, so that I can still use the nic for connections...) Zoltan -- == Geograph (Pty) Ltd P.O. Box 31255 Tokai 7966 Tel:+27-21-7018492 Fax:+27-86-6115323 Mobile: +27-83-6004028 == ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] No channels after starting asterisk
Zaptel Configuration == SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) SPAN 2: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) SPAN 3: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) SPAN 4: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: Individual Clear channel (Default) (Slaves: 01) Channel 02: Individual Clear channel (Default) (Slaves: 02) Channel 03: Individual Clear channel (Default) (Slaves: 03) Channel 04: Individual Clear channel (Default) (Slaves: 04) Channel 05: Individual Clear channel (Default) (Slaves: 05) Channel 06: Individual Clear channel (Default) (Slaves: 06) Channel 07: Individual Clear channel (Default) (Slaves: 07) Channel 08: Individual Clear channel (Default) (Slaves: 08) Channel 09: Individual Clear channel (Default) (Slaves: 09) Channel 10: Individual Clear channel (Default) (Slaves: 10) Channel 11: Individual Clear channel (Default) (Slaves: 11) Channel 12: Individual Clear channel (Default) (Slaves: 12) Channel 13: Individual Clear channel (Default) (Slaves: 13) Channel 14: Individual Clear channel (Default) (Slaves: 14) Channel 15: Individual Clear channel (Default) (Slaves: 15) Channel 16: D-channel (Default) (Slaves: 16) Channel 17: Individual Clear channel (Default) (Slaves: 17) Channel 18: Individual Clear channel (Default) (Slaves: 18) Channel 19: Individual Clear channel (Default) (Slaves: 19) Channel 20: Individual Clear channel (Default) (Slaves: 20) Channel 21: Individual Clear channel (Default) (Slaves: 21) Channel 22: Individual Clear channel (Default) (Slaves: 22) Channel 23: Individual Clear channel (Default) (Slaves: 23) Channel 24: Individual Clear channel (Default) (Slaves: 24) Channel 25: Individual Clear channel (Default) (Slaves: 25) Channel 26: Individual Clear channel (Default) (Slaves: 26) Channel 27: Individual Clear channel (Default) (Slaves: 27) Channel 28: Individual Clear channel (Default) (Slaves: 28) Channel 29: Individual Clear channel (Default) (Slaves: 29) Channel 30: Individual Clear channel (Default) (Slaves: 30) Channel 31: Individual Clear channel (Default) (Slaves: 31) Channel 32: Individual Clear channel (Default) (Slaves: 32) Channel 33: Individual Clear channel (Default) (Slaves: 33) Channel 34: Individual Clear channel (Default) (Slaves: 34) Channel 35: Individual Clear channel (Default) (Slaves: 35) Channel 36: Individual Clear channel (Default) (Slaves: 36) Channel 37: Individual Clear channel (Default) (Slaves: 37) Channel 38: Individual Clear channel (Default) (Slaves: 38) Channel 39: Individual Clear channel (Default) (Slaves: 39) Channel 40: Individual Clear channel (Default) (Slaves: 40) an so on for rest of the channels Tom Hayden wrote: What kind of output do you get with ztcfg -vv ?? -- Tom On 7/13/05, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Wed, Jul 13, 2005 at 05:19:08PM +0200, Kib Eki wrote: Hi, i am running * 1.0.9 with a newer version of the TE405P. Modprobe wct4xxp and ztcfg are OK. zap show channels only shows me the following. my zapata.conf: [pstn] Shouldn't that be [channels] ? Why can't i see or use my channels? -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] No channels after starting asterisk
Bob, there are no error messages. This is the first time we installed a TE405P adapter to the system. So that is the change to the system. Bob Goddard wrote: On Wednesday 13 Jul 2005 16:19, Kib Eki wrote: Hi, i am running * 1.0.9 with a newer version of the TE405P. Modprobe wct4xxp and ztcfg are OK. zap show channels only shows me the following. my zapata.conf: [...] Why can't i see or use my channels? You're not going to get anywhere unless you show us the error messages and what if anything has changed on your system. B ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CONSOLE/dsp
On Wed, Jul 13, 2005 at 05:19:09PM -0700, Barton Fisher wrote: However, I'm not 100% sure the sound card drivers are working. is either chan_alsa.so or chan_oss.so loaded? What kernel version is it and from where? My question is how can I test the sound card separately from Asterisk using only the command line? For example, play a file to sound card. If I know the sound card is working, I should be able find the reason why I can not connect to CONSOLE/dsp The package sox has a handy 'play' utility. It plays to OSS, but also provides an OSS emulation. It can also play gsm files from the sounds/ dir (at least the version on Debian). This should be a standard package on most ditributions. If all seems well but there's no sound, make sure the sound is not muted. I normally use aumix. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Multiple NICs on Asterisk box
On Thu, Jul 14, 2005 at 09:00:04AM +0200, Zoltan Szecsei wrote: Kevin P. Fleming wrote: Zoltan Szecsei wrote: I've traced the problem to be with the firewall and the fact that I have 2 NICs in the box. Now that I have opened port 4569 on both interfaces, asterisk seems happy *but* does anyone know how to force SuSE 9.3 to always bring up a specific NIC before the otherone? SUSE Pro 9.3 assigns IP addresses to NICs based on their MAC addresses, not their slot positions or load order (unless you've done something funky). You should not have any problem with the wrong NIC getting a particular IP address, and Asterisk does not care at all which one is 'eth0'. I wish this were true, but, believe me, on reboot, sometime the pci 3com card gets eth0 and sometime the onboard. Who says computers re-iterate perfectly :-) ? BTW: I need the pci card to be eth0 (sensed first). If you want useful answers from people here, provide some data for people to work with. As for that data: people asked you to look at some specific files. E.g: frankly I still can't tell if both cards get loaded by the same module or by different modules. Frankly, I can't think of a good reason why two different boots of the same machine with very similar config would have different module/card load order. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Wire Tapping on Asterisk
I'm new to asterisk. I would like to ask if there's a feature in asterisk wherein you can monitor ongoing calls, some kinda like tapping into active phone calls? It must have this feature but I do not know where to get some reference to set this up or test this. Can anyone share me some sites as reference? thnx... ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] PRI Q.921 problem
hi guys: we are having Q.921 DLPI problem when trying to establish ISDN layer 2 connection with the E1 switch of the telecom carrier, the carrier switch is unable to establish layer 2 link withour * server with a quad E1 card, Here is a trace provided by telecom. U N DL_RELEASE_IND U N (PRA_PROTOCOL_CONFIG) U N DL_ESTABLISH_REQ look like this DL_RELEASE_IND is part of Q.921, but i don't see this in libPRI, is the LIBPRI implementation missing the DLPI part of the Q.921?I also found some free code of Q.921 Q931 on the net, Linux and FreeBSD seams also contains this DLPI in their isdn drivers. reading the libpri code, it tells it is doing the ua only, no DLPI, shall we add DLPI or we can tell teleco to only use ua layer 2 link establishment? Any hint of help is greatly appreciated. Matt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] auto dialing - call file - channel variable question
Hi When using a call file to place a call I can't seem to figure out how to get the variable alert_info passed to the actual channel (in my case a SIP phone) that an agent is logged in at. Please can someone give me a pointer in the right direction ;) Thanx! Probably best illustrated in an example: Below works great and tells SIP/123 to pick up the call from asterisk then it dials the desired extension: SetVar: _alert_info=auto_answer Channel: SIP/123 Context: autodial Exten: 123456789 Priority: 1 Below doesn't pass alert_info to the sip phone the agent is logged in at, so the agent has to answer the call before the destination extension is dialed: SetVar: _alert_info=auto_answer Channel: AGENT/1001 Context: autodial Exten: 123456789 Priority: 1 Start your day with Yahoo! - make it your home page http://www.yahoo.com/r/hs ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sangoma A104c vs. A104u
Hi, Just a quickie - if I want to implement an * solution purely for voice (well, and physical fax machines / dialup modems..) on EuroISDN E1s, is there any benefit to the A104u over the A104c? I'm just trying to decide if the extra £200 for the A104u is worth it :) Cheers, Gavin. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] No channels after starting asterisk - SOLVED!!!
That what was exactly the mistake in the configuration. I changed [pstn] to [channels] and restartet *. Thank you very much! Tzafrir Cohen wrote: On Wed, Jul 13, 2005 at 05:19:08PM +0200, Kib Eki wrote: Hi, i am running * 1.0.9 with a newer version of the TE405P. Modprobe wct4xxp and ztcfg are OK. zap show channels only shows me the following. my zapata.conf: [pstn] Shouldn't that be [channels] ? Why can't i see or use my channels? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Multiple NICs on Asterisk box
Dave Cotton wrote: On Thu, 2005-07-14 at 09:01 +0200, Zoltan Szecsei wrote: Ken Godee wrote: On reboot sometimes my onboard gigabit nic gets eth0 and sometimes the pci 3COM gets eth0 and this causes havoc with another piece of SW I run. I seem to remember having this type of thing before, only I had two identical NICs, because they were good ones :) Have you tried creating an alias in modprobe.conf (2.6) or modules.conf (2.4) alias eth0 b44 alias eth1 e100 in my case Hi, I'm not sure what you are getting at here. An alias is just a second name for something. If the HW device changes under the original name (eth0), then surely the alias pointing to that original name now too points to the different HW device? Apologies if I have mis-understood you. I seem to think that what is required is a way (at boot time) to make certain that the pci slots get probed before the on M/B nic gets probed, that way ensuring that the pci nic is set up before the on board nic. Can this be done? regards, Zoltan -- == Geograph (Pty) Ltd P.O. Box 31255 Tokai 7966 Tel:+27-21-7018492 Fax:+27-86-6115323 Mobile: +27-83-6004028 == ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] bandwidth of gsm and g729
what are the bandwidths of the gsm codec and g729 codec and are they in same sound quality . Start your day with Yahoo! - make it your home page http://www.yahoo.com/r/hs ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SMS transmit to analog device
Hi all, I want to transmit an SMS to an german analog device T-Sinus 700K which is connected via an a/b-adapter and OctoBRI to my asterisk box. Asterisk 1.0.9-BRIstuffed-0.2.0-RC8j. The SMSC-Nr 0193010 is configured in the T-Sinus. It answers the call and displays SMS is currently transfered. But after hangup no incoming SMS is displayed / reported?! Here is the smsq invocation used: smsq --mttx-channel=Zap/g18/198 --mttx-retries 0 --mttx-callerid 01930100 --oa 041061234567 -t --mr 1 --scts 2005-07-14 10:16:26 Hello Sending SMS from the T-Sinus to asterisk is no problem. Any hints why the SMS is not accepted by the called device are greatly appreciated ;-) Bruno -- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] bandwidth og gsm and g729
what are the bandwidths of the gsm codec and g729 codec and are they in same sound quality . __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] *** install error
On Thu, Jul 14, 2005 at 10:22:03AM +0200, luca vespa wrote: i've installed asterisk yesterday for the first time. i did make and make install for all the directory and after MAKE SAMPLES but when asterisk start i receive message: [app_readfile.so]Jul 14 10:21:17 WARNING[4069]: loader.c:258 ast_load_resource: /usr/lib/asterisk/modules/app_readfile.so: undefined symbol: ast_register_file_version Jul 14 10:21:17 WARNING[4069]: loader.c:440 load_modules: Loading module app_readfile.so failed! how can i resolve it? Left over-modules from a previous installation? Does 'find . -name app_readfile.so' from the source directory produce anything? If not, it probably was not installed from the current 'make install' -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Wire Tapping on Asterisk
Hi,
Yes, there is a way. In extensions.conf, you add a macro as:
[macro-record-on]
exten = s,1,AGI(set-timestamp.agi)
exten = s,2,SetVar(CALLFILENAME=${timestamp}-${ARG2}-${ARG1})
exten = s,3,Monitor(wav,${CALLFILENAME},m)
then, when you want to record the call, you use:
exten = s,1,Macro(record-on,NAME_OF_CHANNEL,${CALLERIDNUM})
this will record to a file named for example
20050704-173558-93xxx-IN.wav (number obfuscated)
The set-timestamp.agi is nothing else than
#!/bin/sh
longtime=`date +%Y%m%d-%H%M%S`
echo SET VARIABLE timestamp $longtime
MAKE SURE OF THE LEGALITY OF DOING THIS IN THE PLACE YOU WILL BE DEPLOYING.
Best regards,
Mike
Christoph wrote:
On Thu, 2005-07-14 at 17:00 +0800, Ian Bert Tusil wrote:
I'm new to asterisk. I would like to ask if there's a feature in
asterisk wherein you can monitor ongoing calls, some kinda like
tapping into active phone calls? It must have this feature but I do
not know where to get some reference to set this up or test this.
Can anyone share me some sites as reference?
As far as I know there is no feature in Asterisk, but I might be wrong.
However, you can use ethereal to tap SIP connections. You simply sniff
the SIP connection and after it's done you can decode it and ethereal
will output a .au file which contains both sides of the conversation.
Also I heared that the Windows tool Cain Able is able to play back
SIP converstaions in real time, but I haven't tested that myself.
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Re: [Asterisk-Users] Multiple NICs on Asterisk box
alias eth0 b44 alias eth1 e100 I seem to think that what is required is a way (at boot time) to make certain that the pci slots get probed before the on M/B nic gets probed, that way ensuring that the pci nic is set up before the on board nic. Can this be done? Hi The aliases in modules.conf should have worked for you, but if you want, you can always do something like: /usr/sbin/rcroute stop /usr/sbin/rcnetwork stop /sbin/rmmod 3c59x /sbin/rmmod eepro100 /sbin/modprobe 3c59x /sbin/modprobe eepro100 /usr/sbin/rcnetwork start /usr/sbin/rcroute start In this case, it forces a 3COM network card to be detected before an onboard Intel EEPRO. You need to do this before you do run any network programs. I'm just not sure that will make much difference, I think the aliases route is the way to go. Regards Hilton Datatex Dynamics CC Web site http://www.datatex.co.za/ Email to [EMAIL PROTECTED] Tel +27215924033 Fax +27215924077 The use of the Datatex e-mail facility is not permitted for the distribution of chain letters or offensive email of any nature whatsoever. Datatex hereby distances itself from and accepts no liability in respect of the unauthorised use of its e-mail facility or the sending of e-mail communications for other than strictly business purposes. Datatex furthermore disclaims liability for any unauthorised instruction for which permission was not granted. Any recipient of an unacceptable communication, a chain letter or offensive material of any nature is requested to report it to [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New Cisco 7960 Firmware 7.5
Kevin How close is 1.2? Year, Months, Days? Andrew On 7/13/05, Kevin P. Fleming [EMAIL PROTECTED] wrote: Andreas Anderson wrote: Does asterisk allready support supervised-transfers-with-correct-number (c sees number of a after b, who transferred a to c, hung up)...? Any other ideas what could be done with RFC3311/Remote-Party-ID-updates? No, not yet. I was working on this last year, but support for RPID headers in many phones I tested (Cisco included) was lacking and/or buggy. It appears that both Cisco and Polycom have improved their firmware in recent months, so I may begin working on this again after we get 1.2 released and we can start making radical changes again... ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- sig Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) WWW: http://lathama.com Email: [EMAIL PROTECTED] - [EMAIL PROTECTED] - [EMAIL PROTECTED] If any of the above are down we have bigger problems than my email! /sig ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SpanDSP rxfax, no tiff
Maybe I over-complicated my question with the mailfax part. If I leave the mailfax step out entirely, then there should be a .tif file, right? But theres not. No tif file gets created at all. Permissions on the fax folder are 777 at the moment. Thanks for the responses so far. /Rob ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Faxing Suggestions
SECOND! I was faced with either trying to get spanDSP to work on a SIP trunk with an old Sharp fax machine (or a fax modem), or signing up with Trustfax. I'm keeping all my documents in PDF anyway, so this seemed like a good option. Cost-wise, they're definitely ahead for me -- a dedicated DID would cost me $2/month at best. While I first thought their UI was a bit awkward, I've recently come to love it as I had to send and receive aout 160 pages in one week. It's nice to be able to keep incoming and outgoing faxes online for re-use. Of all the options out there, this is the least expensive for us low-volume users. Lastly, even though I haven't spent more than $30 there so far, two calls to tech support were answered promptly and courteously. One was an issue on their side which was fixed within an hour, the other was related to a problem on my network. -Original Message- We are using www.trustfax.com ($9.95/yr for an 800 number plus $0.10 per page. Much less expensive then doing anything on your own when one considers total cost of toner, paper, hardware, multi-line hunt, technical support, junk faxes, etc, etc.) There are several others as well. Exactly the same interactions that I've had with them. Nothing but excellent service. I've given up totally on trying to make spandsp work with the TDM04b card for now. I'm not even sure I'd try that approach again even when the TDM card is fixed. The Trustfax.com approach is s much less support intensive for low volume faxing; its almost a no-brainer. The only downside that I've observed (to date) is that sending a Word document to their automated outbound fax interface has a small conversion problem where it doesn't maintain the same margins and text layout. A well-formated document might look completely different when you get the pdf fax image. But, for us that's a very minor issue since 95% of our faxes are inbound. (We still maintain an old analog fax machine for some outbound faxes, but it hasn't been used in weeks.) FWIW, they had never heard of asterisk before; they are now looking into using it for their voice switch. :) Rich ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Panasonic PBX -to- Sirrix BRI: Numbers gettingechoed/duplicated
Hi Peter,
Thanks for your reply.
One possible cause is that Asterisk receives the digits both as isdn
indications (out of band) and as dtmf
Good point ! This sounds like it could be the problem.
Are you sure you have answered the line?
Yes, as far as I know ? In that context I have the following:
[pabx2ip]
exten = s,1,Answer
exten = s,2,Wait(1)
exten = s,3,ResponseTimeout(3)
exten = s,4,Background,enter-ext-of-person
exten = _X.,1,Dial,IAX2/pmb/${EXTEN}
exten = t,1,Hangup
exten = i,1,Goto(s,1)
Should be OK ?
Before the line is answered Asterisk can receive overlap digits
Yea, I thought about this and tried dialtimeout = yes in
/etc/asterisk/sirrix.conf
ref:
http://www.voip-info.org/tiki-index.php?page=Asterisk+config+sirrix.conf
I think the problem is probably, as you pointed out, that Asterisk is
picking up the DTMF stuff as well.
Do you know of a way to disable it ?
Thank you for your help so far - greatly appreciated.
Kindest regards
David Wilson
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- Original Message -
From: Peter Svensson [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, July 14, 2005 10:21 AM
Subject: Re: [Asterisk-Users] Panasonic PBX -to- Sirrix BRI: Numbers
gettingechoed/duplicated
On Thu, 14 Jul 2005, David Wilson wrote:
I have a Panasonic PBX linked to a Sirrix Quad BRI card that is running
in TE (ptp) mode in a Asterisk box - this then links through Internet to
another Asterisk box via IAX2.
When a user on the Panasonic PBX system dials the extension of my Sirrix
Asterisk box, Asterisk answers and says Please dial the number of the
person you are looking for. This is done with cmd Background.
When this user enters an extension number to call the numbers that get
picked up by Asterisk are repeated/echoed.
For example, if a user enters 19 at the voice prompt, Asterisk picks
it up as 1199 and tries to then dial 1199 out to the remote Asterisk
server.
One possible cause is that Asterisk receives the digits both as isdn
indications (out of band) and as dtmf. Are you sure you have answered the
line? On a bri link audio can be passed even without the line being
answered.
Before the line is answered Asterisk can receive overlap digits. While
in overlap reception mode in band (dtmf) digits are ignored. Yuo may want
to enable overlap digits nn the link to the Panasonic.
I am not familiar with this particular BRI card. If it is not based on
zaptel then the configuration will have to be made elsewhere.
Peter
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Re: [Asterisk-Users] Last CVS - High Load
Hi, I upgraded to: Asterisk CVS-HEAD built by [EMAIL PROTECTED] on a i686 running Linux on 2005-07-13 00:40:26 UTC And the MoH was fixed but I had a lost of state of all my SIP devices at the middle of the day yesterday I restarted and waiting to see if the problem reappear. I believe all these problems are corrected now, the major MOH problem patch was reverted and the other problems with device state handling have been fixed as well. FWIW as a datapoint... checked out cvs-head (make update) for zaptel, libpri, and asterisk at about 8:30pm cdt which compiled/installed fine. I placed calls via a cell phone to/from the following itsp's (iax2): teliax.com diamondcard.com Nufone.com and all functioned correctly with dtmf and ivr, etc. No identifiable issues with anything on this simple FC3 system with 19 sip peers, TDM04b, and iax2 itsps. Waiting for recurrance on the lockup (and will likely be waiting forever ;) LoL that last sentence induced me to reply ;) -- Juanjo sin .sig :( ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] how to debug perl agi
Hi,
syslog('info', 'hello Asterisk!');
That should go into the syslog for the facility user. It may end up on
/var/log/messages , /var/log/user.log or whereever your system sends
such log entries.
But why not print to STDERR? IIRC the stderr of AGI scripts goes to the
asterisk console.
Dunno but at least ASTCC as lot of prints to STDERR but none of
these appeared on my console.
BTW I had to patch last ASTCC CVS since it wasn't getting the call time:
--- /home/juanjo/voip/astcc/astcc.agi 2005-07-11 03:28:06.0 -0400
+++ astcc.agi 2005-07-12 01:48:41.0 -0400
@@ -329,9 +329,10 @@
sub calccost() {
my ($adjconn, $adjcost, $answeredtime, $increment) = @_;
- eval { my $adjtime = int(($answeredtime + $increment - 1) /
$increment) * $increment };
+ my $adjtime = int(($answeredtime + $increment - 1) /
$increment) * $increment;
my $cost;
I'm using Perl 5.8:
[EMAIL PROTECTED]:~$ perl -v
This is perl, v5.8.4 built for i386-linux-thread-multi
Copyright 1987-2004, Larry Wall
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Re: [Asterisk-Users] Any suggestions for an IP phone? TFTP fixed
As others have already stated, be careful with assumptions using tftp. You'll chase your tail off trying to figure out why certain things don't work as expected. Very strongly suggest ftp instead. :) Hi, all Stupid me! Under RH (FC3) tftp server is part of xinet. So, I have enabled the tftp server and set all up and I forgot to restart xinet! Dough! Now I am having fun setting up phone. Rudolf - Original Message - [EMAIL PROTECTED] wrote: Polycom does not support Asterisk. Thsi does not mean phones do not work with it. Rudolf P.S. I am having troubles setting up Polycom 300 with tftp server. By some reason phones always say can not contact boot server. Phones are set to use tftp and correct boot server IP is set via dhcp. I will investigate further, but any suggestions are appreciated. I always use FTP instead, it works famously. Make sure you configure the ftp server in DHCP or in the ftp servers settings, as an IP of course, and that you change the ftp password to the password for the user PlcmSpIp on the server. After that it's flawless. Polycom does not support Asterisk. Polycom, the company, does not support the use of the phones with Asterisk. Who cares? SIP is a standard, we don't need any help from them and we don't need their blessing. The phones are excellent quality and work very well with Asterisk, there's no support issue. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Any suggestions for an IP phone?
AAH 1.3, Digium 4-port FXO card connected to PSTN
I am having problems with outbound calls, where the call goes to either an
error message from the PSTN, or a fax number, or a wrong number. It works
correctly maybe 1 time in 10.
Also, outbound calls *sometimes* work if they are numbers previously dialed.
I've pasted below the relevant parts from extensions-additional and
zapata.conf, as well as a bad call log.
Inbound calls work fine. What's wrong with outbound?
Have you tried inserting a w in the dial string?
If not, try something like this:
exten = _770,1,Dial(Zap/g1/w${EXTEN})
Some central office switches don't like the speed at which asterisk
starts sending dtmf. The w inserts a small delay before dialing.
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[Asterisk-Users] No more sound on MOH after adding TE405P
Hi, after we successfully installed the TE405P card (thanks to this list) the musiconhold does not work anymore. Asterisk starts the mpg123 programm but there is no sound we can hear. Thanks, Kib ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Faxing Suggestions
On Thu, 2005-07-14 at 07:01 -0600, Rich Adamson wrote: I've given up totally on trying to make spandsp work with the TDM04b card for now. I'm not even sure I'd try that approach again even when the TDM card is fixed. See my message from a couple of days ago where I had a perfectly working 33.6kbps modem connection: Windows 2000 Server - External Modem - TDM FXS port - asterisk - TE410p - E1 PRI - some dial-up ISP The modem is only a 33.6kbps modem, but I might try and scrounge up a 56kbps modem and plug it in to see how well that works... So, the real question would be, what is so different between your machine and mine ?? Could it be that I used a server class motherboard (needed 3.3V PCI slot) Could it be that I used SATA drives? Could it be that I don't run any services other than the bare minimum? Could it be that the machine isn't really that busy? (only 25 extensions, with generally avg 1 or 2 calls at any time). Could it be luck? Or, maybe it is a combination of all of these things... Short answer, it would seem that it is possible to get it to work... PS, I suppose I should try to make a connection and see if I can keep the connection 'up' for a few days as a single call. That would then show that it really is quite reliable ie, I could have just been lucky for that 2 minutes that I left the connection up for Feel free to ask me questions on my config/setup, or to run further tests... Regards, Adam -- -- Adam Goryachev Website Managers Ph: +61 2 9345 4395[EMAIL PROTECTED] Fax: +61 2 9345 4396www.websitemanagers.com.au ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Any suggestions for an IP phone? TFTP fixed
On Thu, 2005-07-14 at 07:27 -0600, Rich Adamson wrote: As others have already stated, be careful with assumptions using tftp. You'll chase your tail off trying to figure out why certain things don't work as expected. Very strongly suggest ftp instead. :) Well, apparently they work the same for tftp and ftp with newer bootrom now... but I would still prefer FTP ... Also, apparently they also support https, which I would prefer even more, but I haven't tried it as yet... (I think this only works on the 301/501 and 600 as well)... Regards, Adam -- -- Adam Goryachev Website Managers Ph: +61 2 9345 4395[EMAIL PROTECTED] Fax: +61 2 9345 4396www.websitemanagers.com.au ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco 7960 on Asterisk?
Hello, I am just building my first Asterisk server. Looking for a couple of good quality ip phones. I like the Cisco 7960, are they easy to configure to work with Asterisk? What are the alternatives, with a good speaker phone, and simple clean and stylish look like the Cisco? I appreciate your advice. Kind regards Michael Felder IT Medic Australia Pty. Ltd. P: 03 9557 2213 F: 03 9557 2214 M: 0419 568 217 E: [EMAIL PROTECTED] http://www.ITMedic.com.au Keeping your computer systems healthy. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SMS over SIP and Asterisk ??
Well, I mean Instant messaging between two SIP users registered on Asterisk-sip server. The thing is, some sip phones supports instant messaging but, how can I get this feature work in asterisk ? Angel Date: Wed, 13 Jul 2005 20:17:22 -0400 From: Shidan [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] SMS over SIP and Asterisk ?? To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1 Do you mean SMS or a a SIP MESSAGE, the only sure way I can think of to send SMS with SIP, and I'm no SIP expert here, is if there was an SMS MIME type and you just used SIP for the transport, and even if there is such a type I doubt anyone has implemented anything for it yet, let alone *. As to does Asterisk support MESSAGE requests with a plain/text MIME type, you can use the ap. SendText() , look it up on the wiki Regards, Shidan http://www.nuovotel.com On 7/13/05, Angel Diaz [EMAIL PROTECTED] wrote: Hi, Is there a way to send and receive SMS over SIP protocol with Asterisk ? I mean, between two SIP phones like below... SIP_phone A (sending sms) Asterisk SIP_phone B (receiving sms) ...Is it possible ? If so, how could I do it ? Thanks, Angel. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Asterisk and Dell SC420 Server
calvis wrote: Thanks for pointing this out. In my first attempts with Asterisk I was trying to configure a DELL SC400 machine. To my dismay I could never get it working right. I am not the most technical person in the world so I just assumed that Asterisk configuration was beyond my skill level. Now it is dawning on me that it might have been a hardware problem all along. With this new revelation I want to play around with Asterisk again. Could someone please point me to the 'Approved Hardware List'? Thanks, Unfortunately, Digium doesn't provide such a list. All they provide is a very short list of hardware that is known not to work, and support's answer for other hardware that doesn't seem to work is try another motherboard The TDM400 in particular doesn't seem to work with many motherboards that are supposed to be PCI 2.2 Regardless, any Digium board can't stand sharing an interrupt, so make sure you have good control of that in the BIOS. John Novack ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Wire Tapping on Asterisk
Naw, you're wrong. Look at the Monitor command: http://www.voip-info.org/wiki-Asterisk+cmd+monitor -Original Message- From: Christoph [mailto:[EMAIL PROTECTED] Sent: Thursday, July 14, 2005 5:26 AM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Wire Tapping on Asterisk On Thu, 2005-07-14 at 17:00 +0800, Ian Bert Tusil wrote: I'm new to asterisk. I would like to ask if there's a feature in asterisk wherein you can monitor ongoing calls, some kinda like tapping into active phone calls? It must have this feature but I do not know where to get some reference to set this up or test this. Can anyone share me some sites as reference? As far as I know there is no feature in Asterisk, but I might be wrong. However, you can use ethereal to tap SIP connections. You simply sniff the SIP connection and after it's done you can decode it and ethereal will output a .au file which contains both sides of the conversation. Also I heared that the Windows tool Cain Able is able to play back SIP converstaions in real time, but I haven't tested that myself. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] CallerID rewrite php AGI Script
http://muware.com/asterisk -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Thursday, July 14, 2005 12:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] CallerID rewrite php AGI Script Hello all. I am looking for the great callerID rewrite script that does the 411 lookup and then stores the information in a database. If there is information in the Database for the callerid coming in, then use that and pass it along to the phone. I lost my entire system hard drive this week, and slowly rebuilding. This script wasn't in the most recent backup :( :( :( Please help :) thanks. Ben ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SpanDSP rxfax, no tiff
Rob Danz wrote:
If I leave the mailfax step out entirely, then there should be a .tif
file, right? But there’s not. No tif file gets created at all.
Permissions on the fax folder are 777 at the moment.
Are the permissions okay for getting TO that folder? (do you have r+x on
the directories 'above' the fax folder?)
Thanks for the responses so far.
Just trying to help out, I'm using ISDN lines with a DID, for receiving
I do this..
If one calls number X, then Goto(custom-fax,s,1) and then..
[custom-fax]
exten = s,1,Answer
exten = s,2,Macro(faxreceive)
exten = s,3,SetVar(ONZENID=${UNIQUEID})
exten = h,1,system(/usr/local/sbin/mailfax ${FAXFILE}
[EMAIL PROTECTED] ${CALLERIDNUM} ${CALLERIDNAME} ${ONZENID})
[macro-faxreceive]
exten = s,1,SetVar(FAXFILE=/var/spool/asterisk/fax/${UNIQUEID}.tif)
exten = s,2,SetVar(EMAILADDR=${FAX_RX_EMAIL})
exten = s,3,rxfax(${FAXFILE})
exten = s,103,SetVar(EMAILADDR=${FAX_RX_EMAIL})
exten = s,104,Goto(3)
The [macro-faxrecevive] is from AMP (wich I'm using for managing asterisk)
Hope you can do something with this.. :-)
Cheers
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Re: [Asterisk-Users] Cisco 7960 on Asterisk?
Michael, The Cisco phones are excellent. However, configuring your first one will be a bit of a learning curve, but your second, third, etc. will be relatively easy. You may want to consider the 7940 as well, considering that it is cheaper, and that you most likely will not need 6 line appearances on the phone (Then again, you might) Tom On Jul 14, 2005, at 8:59 AM, Michael Felder wrote: Hello, I am just building my first Asterisk server. Looking for a couple of good quality ip phones. I like the Cisco 7960, are they easy to configure to work with Asterisk? What are the alternatives, with a good speaker phone, and simple clean and stylish look like the Cisco? I appreciate your advice. Kind regards Michael Felder IT Medic Australia Pty. Ltd. P: 03 9557 2213 F: 03 9557 2214 M: 0419 568 217 E: [EMAIL PROTECTED] http://www.ITMedic.com.au Keeping your computer systems healthy. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] AgentMonitorOutgoing question
This is my nth post regarding this matter. Where in the dialplan should I put AgentMonitorOutgoing? Can somebody show me how to use it? in extensions.conf: exten = x,1,AgentMonitorOutgoing(c) in agents.conf I set updatecdr to yes. It supposed to put agent/agent id in the channel column in the CDR, but instead it puts SIP/extension number. This link seems to be the only reference: http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+AgentMonitorOutgoing says that something needs to be configured in the agents.conf file, but doesn't specify which part. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cisco 7960 on Asterisk?
Hi Michael, i run several 7960 and 7940 on our network, and they run smooth and without any complications. you will have to upgrade them using the SIP firmware from Cisco (i use versions 7.4 and 7.5 at the moment), you can download them from Cisco's Homepage but you will need a CCO account with a contract for that (or grab then via the e*onkey network, search for P0S3, 0 is zero, not the letter O!!!). You should use SIP on the phone that supports many more features of Asterisk's SIP channel than any other implementation. all is well documented on the wiki on how to upgrade the phone but unfortunately you will have to spend some time in setting up your asterisk and the TFTP-configs for the phone to get it working properly. some links i used in the most early state of getting them to work: http://www.voip-info.org/tiki-index.php?page=Setup%20SiP%20on%207940%20- %207960 http://www.it4u2.com/asterisk2.htm http://www.voip-info.org/tiki-index.php?page=Asterisk+phone+cisco+79xx once you upgraded your phone to SIP firmware, which is the most tricky part of the work, the configuration of Asterisk should be done in a few hours for a newbie to Asterisk. Hope this helps! Regards, Roland -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Felder Sent: Thursday, July 14, 2005 3:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Cisco 7960 on Asterisk? Hello, I am just building my first Asterisk server. Looking for a couple of good quality ip phones. I like the Cisco 7960, are they easy to configure to work with Asterisk? What are the alternatives, with a good speaker phone, and simple clean and stylish look like the Cisco? I appreciate your advice. Kind regards Michael Felder IT Medic Australia Pty. Ltd. P: 03 9557 2213 F: 03 9557 2214 M: 0419 568 217 E: [EMAIL PROTECTED] http://www.ITMedic.com.au Keeping your computer systems healthy. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk number of calls
Good day all What is the amount of calls that asterisk can handle,SIP and from/to PSTN -- Thanks Altus Snyman Stormcorp Network Solutions +27 11 8071141 exten 301 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Plzzzzz tell me how to register users in oh323.conf
Assalam Alaikum i m using oh323.conf n im calling netmeeting/SJPhoneusing Dial(oh323/IP address of netmeeting or anysoftphone) how can i call any extension first of all tell me how to register a uid n password there let's say i've a user type=friend username=adeel secret=adeel context=incoming mailbox=31 plz plz plzz send me just a sample oh323.conf n related portion of extesions.conf containg above info Start your day with Yahoo! - make it your home page ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960 on Asterisk?
much discusions in this forum about phone recommendation seems that favorites are cisco, polycom, sipura, snom, uniden and zultys (alphabetical order ;-) I like vendor independace and maximum interoberability, so polycom, cisco/sipura are not my favorites, also talking to zultys can't be direct, but via partner, there is left uniden and snom, my favorit is currently snom 360 :-) PJ Michael Felder wrote: Hello, I am just building my first Asterisk server. Looking for a couple of good quality ip phones. I like the Cisco 7960, are they easy to configure to work with Asterisk? What are the alternatives, with a good speaker phone, and simple clean and stylish look like the Cisco? I appreciate your advice. Kind regards Michael Felder IT Medic Australia Pty. Ltd. P: 03 9557 2213 F: 03 9557 2214 M: 0419 568 217 E: [EMAIL PROTECTED] http://www.ITMedic.com.au Keeping your computer systems healthy. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] problems with outbound routing
On 7/14/05 6:33 AM, Rich Adamson [EMAIL PROTECTED] wrote:
AAH 1.3, Digium 4-port FXO card connected to PSTN
I am having problems with outbound calls, where the call goes to either an
error message from the PSTN, or a fax number, or a wrong number. It works
correctly maybe 1 time in 10.
Also, outbound calls *sometimes* work if they are numbers previously dialed.
I've pasted below the relevant parts from extensions-additional and
zapata.conf, as well as a bad call log.
Inbound calls work fine. What's wrong with outbound?
Have you tried inserting a w in the dial string?
If not, try something like this:
exten = _770,1,Dial(Zap/g1/w${EXTEN})
Some central office switches don't like the speed at which asterisk
starts sending dtmf. The w inserts a small delay before dialing.
Yes. I have the same problem even with the w in place.
Thanks in advance for any other clues.
dn
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RE: [Asterisk-Users] Zaptel won't compile under Fedora Core 4
My
fault. The kernel and kernel-dev packages got out of sync.
All is better now.
-Original Message-From: Bates, Curtis
Sent: Tuesday, July 12, 2005 2:39 PMTo: Eric Bullen;
Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE:
[Asterisk-Users] Zaptel won't compile under Fedora Core 4
Which version of Zaptel are you using? I am using version 1.09
and having issues, I did not have issues with 1.08.
-Original Message-From: Eric Bullen
[mailto:[EMAIL PROTECTED]Sent: Tuesday, July 12,
2005 12:36 PMTo: Asterisk Users Mailing List - Non-Commercial
DiscussionSubject: Re: [Asterisk-Users] Zaptel won't compile
under Fedora Core 4On 7/11/05, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On
Tue, Jul 12, 2005 at 11:04:16AM +1000, Gonzalo Servat wrote: On
7/12/05, Eric Bullen [EMAIL PROTECTED]
wrote: I hope someone can offer me some help with this.
Basically, the current CVS version of Zaptel will not
compile under Fedora Core 4. I have closely followed the
directions in http://www.voip-info.org/tiki-index.php?page=Asterisk+Fedora+Core+3
using the versions given in the FC4 distro with no
luck.Here's the output when I run "make linux26".
Any help would be great. TIA. [...snip...]
In file included from /asterisk_source/zaptel/zaptel.c:40:
/asterisk_source/zaptel/zconfig.h:10:27:
error: linux/version.h: No such file or
directory Try installing kernel-source and/or
glibc-kernheaders RPMs from the FC4 CDs.kernel-glibcheaders is the
part of the kernel headers that user-space programs need. Not any good
reference for any kernel module to buildwith.linux/version.h
is generated as part of the configuration process of thekernel
configuration process ("make {,menu,x,g}config") of the kernel
source.What version of zaptel do you try to build? For what
kernel version?Your distro's default or your one you've built
yourself?
That did the trick- I untarred the kernel source,
ran "make oldconfig make", and got the linux/version.h file. Then
did a "ln -s kernels/linux-2.6.12 linux-2.6" in the /usr/src dir. Once that
was done, went to the zaptel dir and it compiled beautifully. Thank you so
much.To answer your question, I am running the latest version in CVS
using "cvs update -d -r v1-0", and kernel 2.6.12. I ended up building my own
kernel (not installed) to get this to work (not sure if I could do it the
other way).Thanks again for the help, and hopefully my notes will
help others.-Eric-A.G.
Edwards Sons' outgoing and incoming e-mails are
electronicallyarchived and subject to review and/or disclosure to someone
other than the
recipient.-
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Re: [Asterisk-Users] NO calling tone
On 14:12, Thu 14 Jul 05, Bill Wong wrote:
Thank you Michiel.
I tried to remove m and use r , but still not working, after I change r
to R , it is working. Anybody know why?
This is in the 'show application dial'
'r' -- indicate ringing to the calling party, pass no audio until answered.
'R' -- indicate ringing to the calling partywhen the called party indicates
ringing, pass no audio until answered.
strange the r is not working.
R will not generate ringing sound, it will simply pass the
ringing sound the other end provides.
Michiel van Baak wrote:
On 11:12, Wed 13 Jul 05, Bill Wong wrote:
Can you show me the example, i am newbie.NOt sure whether the code i
modified is correct or not..
my code as below..
exten = 671042,1,Dial(${PHONES1},20,Ttmr)
loose the m.
m = provide music while ringing
r = provide ring sound while ringing.
Using both is conflicting and will result in silence while
ringing.
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[Asterisk-Users] problems with g729
when i display g729 on iax.conf and make a call using g729 it gives this in several lines: Jul 14 17:30:58 WARNING[14196]: codec_g729.c:180 g729tolin_framein: Out of G.729 Decoder Licenses! Start your day with Yahoo! - make it your home page http://www.yahoo.com/r/hs ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Multiple NICs on Asterisk box
Tzafrir Cohen wrote: If you want useful answers from people here, provide some data for people to work with. As for that data: people asked you to look at some specific files. E.g: frankly I still can't tell if both cards get loaded by the same module or by different modules. Frankly, I can't think of a good reason why two different boots of the same machine with very similar config would have different module/card load order. OK - time to summarise so that I dont end up with bitty answers and attempts from all the various suggestions I have. Tzafrir - your accusation in your first paragraph above is correct - I missed your pointer to the HOW-TO and I threw away the PERSISTENT_NAME idea as intuitively I didnt think it would work. Apologies to all. Now, lets go through it in this order. 1) I need to show conclusively what is going funny. 2) The PERSISTENT_NAME idea 3) The alias suggestion 4) The manual rmmod/modprobe idea Lets go: +++ 1) I need to show conclusively what is going funny. OK, till now the machine was up since yesterday. I copied the /var/log/boot.msg file and powered off on again (not init 6). As luck would have it, the NICs swopped around, so here are the exerpts from bot boot.msg file. (No, don't be cynical - I haven't just created these entries in an editer :-) ) = exerpt Start Unicode mode doneLoading console font lat9w-16.psfu -m trivial G0:loadable Waiting for zap to come online ...OK doneSetting up network interfaces: lo loIP address: 127.0.0.1/8 doneeth0 device: Realtek Semiconductor Co., Ltd. RTL-8169 Gigabit Ethernet (rev 10) eth0 configuration: eth-id-00:11:09:7f:15:70 eth0 IP address: 192.168.0.100/24 doneeth1 device: 3Com Corporation 3c905C-TX/TX-M [Tornado] (rev 74) eth1 configuration: eth-id-00:50:da:df:4a:3f eth1 IP address: 192.168.1.100/24 doneSetting up service network . . . . . . . . . . . . . . . .done = exerpt Start Unicode mode doneLoading console font lat9w-16.psfu -m trivial G0:loadable doneSetting up network interfaces: lo loIP address: 127.0.0.1/8 doneeth0 device: 3Com Corporation 3c905C-TX/TX-M [Tornado] (rev 74) eth0 configuration: eth-id-00:50:da:df:4a:3f eth0 IP address: 192.168.1.100/24 doneeth1 device: Realtek Semiconductor Co., Ltd. RTL-8169 Gigabit Ethernet (rev 10) eth1 configuration: eth-id-00:11:09:7f:15:70 eth1 IP address: 192.168.0.100/24 doneSetting up service network . . . . . . . . . . . . . . . .done OK - now you can see what's happening (I've removed the init.d/zaptel entry which is why it is not occuring the second time) +++ 2) The PERSISTENT_NAME idea I first loaded PERSISTENT_NAME=eth2 (and eth3) in the two files so that I could check if my software was checking against eth0 (in which case it would have failed as only eth2 and eth3 exists) - and (as luck would have it) this time the 3COM card initialised first, and the software authenticated correctly. This means to me that the eth0 name is irrelevant and the SW authenticates against the MAC address of the1st nic initialised. So, I need not worry about the interface name, but I do have to lock down the 3COM nic so that it initialises first. +++ 3) The alias suggestion I did not understand this at the time I received it - Had I noticed Tzafrir's pointer to the ethernet HOWTO, I would have realised that alias in this context was not giving an alias to the eth0/1 names. Oops sorry, again. However, although I discarded this idea originally for the wrong reason, it still shouldn't (and didn't) solve the problem. If the names eth0 eth1 are defined only after the NICs are sensed, then force-loading the wrong module should not force the interfaces to swop around. I added the 2 alias entries into /etc/modprobe.conf.local and did an init 6. The darn 3com came up first so nothing was proven (although the hopes ran higher). I powered down and powered up and luckily the Realtek came up first - and as eth0. I now expected the 3c59x module to be forced onto the Realtek interface (due to the alias's), but there were no boot time or other messages and I could ping the network - but not authenticate my SW as the 3com NIC was now 2nd in the que. +++ 4) The manual rmmod/modprobe idea ##/usr/sbin/rcroute stop# locate rcroute showed nothing for my SuSE 9.3 /sbin/rcnetwork stop /sbin/rmmod 3c59x /sbin/rmmod r8169 /sbin/modprobe 3c59x /sbin/modprobe r8169 /sbin/rcnetwork start ## /usr/sbin/rcroute start Brilliant. This does exactly what I need. Thanks to Hilton Williams for hitting the nail on the head - but thanks also to all who endured the 12 (now
[Asterisk-Users] Changing the voice in Asterisk
Has anyone had any luck in changing the voices for Festival and Asterisk? I have Festival installed and working, but can not get the voice different from the default. Thanks, Jason Jason-- Assuming you follow the installation instructions, and install the Mbrola and other goodies for all the possible different voices, then you can, while running festival in a terminal window, run the following commands (in 195): (SayText This is the default voice. It sounds like K.E.D. Diphone?) (tts text_to_read nil) (voice_cstr_us_awb_arctic_multisyn) (SayText This is AWB arctic multisyn) (tts text_to_read nil) (voice_cstr_us_jmk_arctic_multisyn) (SayText This is JMK arctic multisyn) (tts text_to_read nil) (voice_el_diphone) (SayText This is E.L. Diphone) (tts text_to_read nil) (voice_kal_diphone) (SayText This is K.A.L. Diphone) (tts text_to_read nil) (voice_rab_diphone) (SayText This is R.A.B. Diphone) (tts text_to_read nil) (voice_don_diphone) (SayText This is D.O.N. Diphone) (tts text_to_read nil) (voice_ked_diphone) (SayText This is K.E.D. Diphone) (tts text_to_read nil) (voice_us1_mbrola) (SayText This is mbrola U.S. 1) (tts text_to_read nil) (voice_us2_mbrola) (SayText This is mbrola U.S. 2) (tts text_to_read nil) (voice_us3_mbrola) (SayText This is mbrola U.S. 3) (tts text_to_read nil) (voice_en1_mbrola) (SayText This is mbrola E.N. 1) (tts text_to_read nil) If you are able to decide on a particular voice, you can make it the default by including your lines in the lib/siteinit.scm file: (set! voice_default 'voice_cstr_us_awb_arctic_multisyn) (provide 'siteinit) My advice is to play around with the system and read the docs. There is no substitute. I've published my WhoIsIt-1.1.tar.gz tarball with some scripts to generate files from festival, for all the country names, and the physical location of the different area codes in the US, etc. murf -- Steve Murphy [EMAIL PROTECTED] Electronic Tools Company smime.p7s Description: S/MIME cryptographic signature ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SMS in Belgium
Hiya,
I've been doing some testing with SMS in Belgium (Belgacom), sending SMS
seems to work fine. (with .call file and a context that handles the sending)
The problem is however, receiving.. Hardware and software is quadBRI
from Junghanns, bristuff RC8h (asterisk 1.0.8), Debian 3.1.
For receiving SMS in asterisk, I am using:
[custom-smsrx]
exten = s/0171701,1,Verbose(Receiving SMS from ${CALLERIDNUM})
exten = s/0171701,2,Answer
exten = s/0171701,3,Wait(1)
exten = s/0171701,4,SMS(from-pstn,a)
exten = s/0171701,5,Hangup
I do get some logging that shows me it seems to work, but nothing gets
written to the /var/spool/asterisk/sms dir.. Here's the log:
Jul 14 15:17:30 VERBOSE[12055]: -- Executing Goto(Zap/5-1,
custom-smsrx|s/0171701|1) in new stack
Jul 14 15:17:30 VERBOSE[12055]: -- Goto (custom-smsrx,s/0171701,1)
Jul 14 15:17:30 VERBOSE[12055]: -- Executing Verbose(Zap/5-1,
Receiving SMS from 0171701) in new stack
Jul 14 15:17:30 VERBOSE[12055]: -- Executing Answer(Zap/5-1, )
in new stack
Jul 14 15:17:30 WARNING[12055]: Unable to request echo training on channel 5
Jul 14 15:17:30 VERBOSE[12055]: -- Executing Wait(Zap/5-1, 1) in
new stack
Jul 14 15:17:31 DEBUG[12055]: Scheduling timer at 160 sample intervals
Jul 14 15:17:31 DEBUG[12055]: Generator got voice, switching to phase
locked mode
Jul 14 15:17:31 DEBUG[12055]: Scheduling timer at 0 sample intervals
Jul 14 15:17:33 VERBOSE[12055]: -- SMS RX 91 8D 00 0A 81 20...
Jul 14 15:17:33 VERBOSE[12055]: -- SMS TX 95 02 00 00 69 00...
Jul 14 15:17:35 VERBOSE[12055]: -- SMS RX 91 8D 04 0A 81 20...
Jul 14 15:17:35 VERBOSE[12055]: -- SMS TX 95 02 00 00 69 00...
Jul 14 15:17:36 VERBOSE[12055]: -- SMS RX 94 00 6C 0A 81 20...
Jul 14 15:17:36 VERBOSE[12055]: -- Executing Hangup(Zap/5-1, )
in new stack
Any idea.. anyone?
Cheers!
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RE: [Asterisk-Users] asterisk number of calls
I would think that question would be as silly as me asking you a) how many people can I fit in a vehicle or b) how many web users could I have access my apache web server... Need more details to make that judgement. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of altus Sent: Thursday, July 14, 2005 9:38 AM To: asterisk Subject: [Asterisk-Users] asterisk number of calls Good day all What is the amount of calls that asterisk can handle,SIP and from/to PSTN -- Thanks Altus Snyman Stormcorp Network Solutions +27 11 8071141 exten 301 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Robert A. Huddleston.vcf Description: Binary data ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk number of calls
Could someone also tell me how much a car costs ? What i mean is, it all depends on your server and the codecs used, the max is currently a DS3 worth of calls. altus wrote: Good day all What is the amount of calls that asterisk can handle,SIP and from/to PSTN signature.asc Description: OpenPGP digital signature ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] HFC + DECT sync
Hi, I have a proprietary DECT PBX connected to a HFC-S card. As soon as I plug the PBX into the HFC card, some of the DECT phones lose their sync: They switch between no connection and connected every few seconds. When I unplug ISDN, the phones are working again. So the PBX seems to generate some DECT timing signal from the ISDN bus. Is there a way to tweak the parameters of the HFC card so the DECT timing is correct? Could this problem be avoided by using a professional 4-port HFC card? Andreas ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk adit 600 with CMG card
Folks: Does anybody has information how to configure an adit 600 w/ a CMG card. I found some info in google but wasn't clear and not working. Any information you can provide will be really appreciated. Regards, J.Raborg ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MeetMe + CONSOLE
Hi all, Can anyone help me to make my soundcard (CONSOLE) to participate in a meetme room automatically from my dialplan. I want the soundcard to join a meetme room when someone else joins the room. Thanks a lot! == Eduardo J. López Martínez [EMAIL PROTECTED] Isabel Operation Center [EMAIL PROTECTED] DIT - Dept. Ing. Sist. Telemáticos Tlf: +34 91 3367366 (3036) UPM - Univ. Politecnica de Madrid Fax: +34 91 3367333 ETSI Telecomunicacion 28040 Madrid, Spain == ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CVS HEAD voicemailbox full error
yup. had exactly the same problem as you. been spending the last hour trying to figure out what I did wrong in my config. guess how I fixed it? cd /usr/src/asterisk cvs update make install simple really! ;-) I guess someone posted a bugfix a few mins ago and I just picked it up! ;-) cheers, Mark On 7/14/05, Vahan Yerkanian [EMAIL PROTECTED] wrote: Anyone else has problems with CVS HEAD's from today with voicemailhanging up silently without any debug/error messages when checked? It also keeps insisting that the user's voice mailbox is full and can'tstore more messages even if I clear/rebuild the/var/spool/asterisk/voicemail stuff.I've tried falling back to voicemail.conf entries from realtime voicemail with the same result.Thanks,Vahan___Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- regards,Mark P. EdwardsFWD: 667917 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk number of calls
Zoa wrote: Could someone also tell me how much a car costs ? What i mean is, it all depends on your server and the codecs used, the max is currently a DS3 worth of calls. Ah - glad you clarified - I thought it depended on the make model of car you wanted :-) Z altus wrote: Good day all What is the amount of calls that asterisk can handle,SIP and from/to PSTN ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- == Geograph (Pty) Ltd P.O. Box 31255 Tokai 7966 Tel:+27-21-7018492 Fax:+27-86-6115323 Mobile: +27-83-6004028 == ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960 on Asterisk?
An easier route is to have the vendor upgrade to SIP (usually for an additional charge. I've seen from $5 to $30) That way you don't have to get a contract, you don't have to find the files, and you don't have to fiddle with the firmware to get it to work. I have done both, and I think that if you can get the vendor to do it for $5-$10US, it's worth it. Tom On Jul 14, 2005, at 9:39 AM, Roland Zagler wrote: Hi Michael, i run several 7960 and 7940 on our network, and they run smooth and without any complications. you will have to upgrade them using the SIP firmware from Cisco (i use versions 7.4 and 7.5 at the moment), you can download them from Cisco's Homepage but you will need a CCO account with a contract for that (or grab then via the e*onkey network, search for P0S3, 0 is zero, not the letter O!!!). You should use SIP on the phone that supports many more features of Asterisk's SIP channel than any other implementation. all is well documented on the wiki on how to upgrade the phone but unfortunately you will have to spend some time in setting up your asterisk and the TFTP-configs for the phone to get it working properly. some links i used in the most early state of getting them to work: http://www.voip-info.org/tiki-index.php?page=Setup%20SiP%20on% 207940%20- %207960 http://www.it4u2.com/asterisk2.htm http://www.voip-info.org/tiki-index.php?page=Asterisk+phone+cisco+79xx once you upgraded your phone to SIP firmware, which is the most tricky part of the work, the configuration of Asterisk should be done in a few hours for a newbie to Asterisk. Hope this helps! Regards, Roland -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Felder Sent: Thursday, July 14, 2005 3:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Cisco 7960 on Asterisk? Hello, I am just building my first Asterisk server. Looking for a couple of good quality ip phones. I like the Cisco 7960, are they easy to configure to work with Asterisk? What are the alternatives, with a good speaker phone, and simple clean and stylish look like the Cisco? I appreciate your advice. Kind regards Michael Felder IT Medic Australia Pty. Ltd. P: 03 9557 2213 F: 03 9557 2214 M: 0419 568 217 E: [EMAIL PROTECTED] http://www.ITMedic.com.au Keeping your computer systems healthy. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] problems with g729
wassim darwish wrote: when i display g729 on iax.conf and make a call using g729 it gives this in several lines: Jul 14 17:30:58 WARNING[14196]: codec_g729.c:180 g729tolin_framein: Out of G.729 Decoder Licenses! Well, if that isn't the most self-explanatory error in the entire asterisk code, I don't know what is. -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Systems Admin; Telecom Newbie - What do I need?
Thanks for all the great help! I finally feel a little grounded when thinking about this stuff... at least enough to put a ballpark figure on my budget, anyway. But as I think about it a little more, one huge question appears to me... do I need to massively expand my network? I had suggested that I would like to use soft phones on my Macs, and several people here (and elsewhere) have mentioned that that technology isn't really ready for primetime... not for serious business applications anyway. But currently, I only have one ethernet jack per office. Routing another 60 or so ports would add a very substantial expense in both cabling and backbone expansion (what category ethernet is required, BTW?). My ComDial routes over what appears to be 4-wire phone wire RJ- whatever... 11? 45? I get those confused. Anyway, are those wires acceptable? What do you folks do in a situation like mine? And is there any chance in hell I could use my ComDial DigiTech 7700 phones with Asterisk? I assume that's right out, but might as well ask ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Polycom behind firewall issue
I have a user that just got a broadband connection so she could have an extension off our pbx. The service is DSL and uses a speedstream 5200 dsl router. I sent her a Polycom IP300. At first it would not access the config files via ftp so I had tech support walk her through setting the phone's internal IP to be the dmz. This allowed me to set up the phone using the web interface and now it registers. We had NAT problems so I set the NAT features of the phone: IP Address: 67.136.nnn.nnn Signalling Port : 5060 Media Port Start: 1 In sip.conf, I have nat=yes externalip=67.136.nnn.nnn qualify=yes I can call the user and she can hear me. If she calls me, no voice can be heard either way. When I run sip show channels, I see: Peer User/ANRCall ID Seq (Tx/Rx) Format Last Msg 192.168.0.169805 02fe2b2a684 00102/0 g729Tx: ACK 67.136.nnn.nnn893 8dae34ea-ae 00101/1 g729Rx: INVITE 67.136.nnn.nnn(None) 3926de51-a1 00101/1 unknow Rx: REGISTER and it just stays like that until the call is terminated. I would think it was an rtp / nat problem, any ideas how to fix? -- Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 Cell: 264-235-5670 Yahoo IM: [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Business Edition
Andre Lepage wrote: any body know the real difference between the BE and the free one? The 'real' difference? What do you mean? The website is pretty clear on this topic, as are the multitude of previous threads in the mailing list archives where we've talked about this. Asterisk Business Edition is a snapshot of the development tree, with some features removed and license control added. It's been tested, documented and comes with installation and technical support. It does not contain any features or bug fixes that are not in the open source version. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CVS HEAD voicemailbox full error
I just copied an older app_voicemail.so from another * box. :) Mark Edwards wrote: yup. had exactly the same problem as you. been spending the last hour trying to figure out what I did wrong in my config. guess how I fixed it? cd /usr/src/asterisk cvs update make install simple really! ;-) I guess someone posted a bugfix a few mins ago and I just picked it up! ;-) cheers, Mark On 7/14/05, *Vahan Yerkanian* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Anyone else has problems with CVS HEAD's from today with voicemail hanging up silently without any debug/error messages when checked? It also keeps insisting that the user's voice mailbox is full and can't store more messages even if I clear/rebuild the /var/spool/asterisk/voicemail stuff. I've tried falling back to voicemail.conf entries from realtime voicemail with the same result. Thanks, Vahan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com mailto:Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- regards, Mark P. Edwards FWD: 667917 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users begin:vcard fn:Vahan Yerkanian n:Yerkanian;Vahan org:ARMINCO Global Telecommunications;Head, Research Development dept. adr:;;28, Isahakian ave., PO BOX 10;Yerevan;;375009;Armenia email;internet:[EMAIL PROTECTED] x-mozilla-html:FALSE url:http://www.arminco.com/ version:2.1 end:vcard ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CVS HEAD voicemailbox full error
Vahan Yerkanian wrote: Anyone else has problems with CVS HEAD's from today with voicemail hanging up silently without any debug/error messages when checked? There was a locking bug in app_voicemail that was fixed yesterday afternoon (CDT). Please ensure that you are running an up-to-date copy before reporting problems... and if you are still having this problem with current CVS HEAD, open a bug in Mantis and get a proper call trace so we can see what is happening. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Systems Admin; Telecom Newbie - What do I ne ed?
But currently, I only have one ethernet jack per office. Routing another 60 or so ports would add a very substantial expense in both cabling and backbone expansion (what category ethernet is required, BTW?). Most decent phones have an ethernet passthrough (2 port) so you can plug in your PC. As long as your LAN is decent (Cat5 100baseT switched) the overhead using VoIP is negligible. I have used the 3Com NJ wall jacks with good success: http://www.3com.com/products/en_US/detail.jsp?tab=featurespathtype=purchase sku=3CNJ90 It's basically a 4 port switch that you replace your wall jack with. I used the NJ200, it allows you to set priority per port, although I think they are discontinued now. In combination with a 3Com power over Ethernet injector, I was able to expand a 24 port LAN to a 96 port LAN with a per-port cost of $62 Cdn. And, 24 ports of those 96 are PoE, so I can plug my phones right in to port 1 and they power up, no external power supply needed. hth ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Systems Admin; Telecom Newbie - What do I need?
On Thu, 2005-07-14 at 11:20 -0400, Ed Pastore wrote: But currently, I only have one ethernet jack per office. Routing another 60 or so ports would add a very substantial expense in both cabling and backbone expansion (what category ethernet is required, BTW?). Use a phone like the polycom IP301/501/600 which has a built-in 2 port 10/100 switch. ie, take the existing cable and plug it into the phone, then take a second cable, connect one end to the phone, and the other to your PC. No need for any additional major investment My ComDial routes over what appears to be 4-wire phone wire RJ- whatever... 11? 45? I get those confused. Anyway, are those wires acceptable? What do you folks do in a situation like mine? That is likely RJ11 connections, using cat3 cable... (ie, standard telephony cable)... Apparently you can run 10Mbps over cat3, but it has been said that it isn't the best thing to try and do (I've never done it, nor do I know much about this stuff). And is there any chance in hell I could use my ComDial DigiTech 7700 phones with Asterisk? I assume that's right out, but might as well ask Well, there might be some way you could add a PRI connection from asterisk to your comdial system, then configure your comdial system so that all 'special' service numbers, and external calls are routed to asterisk... Then there is probably a big battle to try and configure everything to forward to the right system to support all the features you want from asterisk etc.. (ie, there must be a reason you are thinking of replacing the existing system?). The short answer is no :) or get a consultant in to talk to you... Mainly you should take note of the first answer above about VoIP phones with dual ethernet switch... I think that is what you are looking for... Regards, Adam -- -- Adam Goryachev Website Managers Ph: +61 2 9345 4395[EMAIL PROTECTED] Fax: +61 2 9345 4396www.websitemanagers.com.au ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Any suggestions for an IP phone?
Pavel Jezek wrote: according to this debate, I would like to try snom 360 still more (features, opensource support, linux based) ;-) any good or bad experience with support from snom? or reliability of snom phones? PJ I've been fiddling with a set of Snom 360's for a while now and really the worst I can say about them are: 1) The buttons feel. Odd. They seem to be wearing in, though. When I first got them they were a bit stiff/unresponsive, but my main testing phone is nicely broken in by now. 2) Occasionally my phone's display has gotten garbled. But I'm fast and crazy and running with the beta firmware. Some of the stuff in 3.60k beta made subscribe/notify seem to work better, but both 3.60k and q both garbled the screen. 3.60l seems stable, though. The worst is the occasional inexplicable screen clearing events. It _seems_ like the phone is still fine, but it has forgotten about the screen entirely. Again, I think that's a beta firmware issue. 3) Related, Snom releases new firmwares for free on a fairly regular basis. Which is good and bad. Read it as you will. 4) Snom seems to pay attention to this mailing list, they've answered at least one of my questions already. 5) The screen seems.. Underutilized. I mean, right now I have 4 button labels, a big analog clock and date, my line appearance and a slightly goofy snom.com logo. Incoming calls do a little song and dance, but it seems like you could do more with the display and rely less on the hard lights. OTOH there would be an application break from the 190 firmware for doing this. 6) There are still the odd little corners where there's some polish missing - e.g. usually the display indicates which button mode you're in (abc, ABC or 123), but you find yourself in places where it doesn't. Usually after you've changed modes to deal with the occasional password issue. 7) Odd personal complaint, but snom hasn't learned the trick of tucking a pound of iron away in the base of the phone to make it seem more sturdy that I like out of telecomm products. 8) Memory? I've started seeing low memory warnings with 2 line appearances and under 30 phone book entries. (fortunately project Ghetto Queue failed to work and I went back to a single line..) For the most part I'm really happy with them, though. There's a learning curve, but what doesn't have one? I say this as someone who hasn't touched any other hardphones, though, so take it with a grain of salt. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IP Phone with Standard Power Ethernet
Interesting, since Netgear doesn't mention that, but it now makes sense. On 7/13/05, Kevin P. Fleming [EMAIL PROTECTED] wrote: C F wrote: According to them it doesn't work and you are right, however I have gotten it to work with the follwoing: http://www.netgear.com/products/details/FSM7326P.php Other posters have stated that switch supports both 802.3af and Cisco proprietary power protocol, so it would work without a special cable in Cisco mode. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Unable to call certain 800 numbers through Teliax
After all of your feedback and a discussion at Teliax we have fixed this issues. It appears that when dialing a PSTN number, using the 'r' option is really unnecessary. Furthermore, some IAX clients and older phones (e.g. Cisco 20 VIP) require us to Answer() the call before dialing the PSTN network or Teliax. For more information, see the thread on Teliax at http://www.teliax.com/forum/viewtopic.php?p=544#544. Thanks for all the help! Cullin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Wednesday, July 13, 2005 12:38 PM To: asterisk-users@lists.digium.com Cc: asterisk-users@lists.digium.com; [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Unable to call certain 800 numbers through Teliax We are unable to call certain 800 numbers through Teliax but I thought I would post this here and see if anyone else had the same problem with either Teliax or other carriers. The 800 numbers causing problems pick-up the call right away and are in the US - American Airlines (8004337300) and Staples (800-378-2753) - we can call many other 800 numbers just fine. My users have reported the same problem with AA, we also use Teliax. I coul care less about Staples but American Airlines is the airline that serves this destination, so it is important to us. I'm not the OP, but I tested both the AA and Staples numbers again this morning via teliax. Still working just fine here (C7960, cvs-head from last night). So, if its not working for both of you, the problem must be: - already fixed in asterisk head, or, - the iax2 call termination equipment (not necessarily asterisk) used by teliax to complete your calls is different from my calls. We use teliax and I had a similar problem with UPS. I can currently call Staples and AA fine. The problem was with numbers that did not generate a ring tone before answering. I solved this problem by changing my Dial command for outbound. I had the 'r' option in there before, so essentially the number would just keep ringing to the user, while on the other end it had actually answered. If this is not your problem, please specify in more detail the behavior you are seeing. What is the output on the asterisk console when one of these calls is made? What version of Asterisk are you using? -Ron ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Systems Admin; Telecom Newbie - What do I need?
Ed Pastore wrote: Thanks for all the great help! I finally feel a little grounded when thinking about this stuff... at least enough to put a ballpark figure on my budget, anyway. But as I think about it a little more, one huge question appears to me... do I need to massively expand my network? You MAY want to also consider a gradual transition away from your Comdial, rather than a full cutover. I had suggested that I would like to use soft phones on my Macs, and several people here (and elsewhere) have mentioned that that technology isn't really ready for primetime... not for serious business applications anyway. And some would argue that also applies to Asterisk. Many features that are taken for granted on business telephone systems do not yet exist or are difficult to configure in Asterisk, along with a less than optimum POTS line interface. Faxing should bypass it as well. But currently, I only have one ethernet jack per office. Routing another 60 or so ports would add a very substantial expense in both cabling and backbone expansion (what category ethernet is required, BTW?). My ComDial routes over what appears to be 4-wire phone wire RJ- whatever... 11? 45? I get those confused. Anyway, are those wires acceptable? What do you folks do in a situation like mine? Depends. Telephone system wiring , depending on when it was installed and by whom, could be 4 wires with no twist, level 1, level 3 or higher wiring. AFAIK, most of the hard IP phones will work on Level 3, but you will need to change the 6 position modular jack out to an 8 position one. ( RJ designations are frequently misused, and refer to wiring patterns defined in Part 68 of the FCC rules ( in the US )) Some later Comdial systems could also use only one pair, and moves and changes over the years could have left LOTS of little gotchas in the wiring. What you really want is a hardphone with a CAt 5 input and a passthrough so you can continue to use your existing wiring, but I don't know if they are made ( yet ) And is there any chance in hell I could use my ComDial DigiTech 7700 phones with Asterisk? I assume that's right out, but might as well ask No chance, unless you retain the Comdial cabinet John Novack ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: asunto_mensaje_entrante
Hasta el día 31 de Julio permaneceré de vacaciones, por lo que cualquier tipo de consulta, técnica o comercial debe redirigirla a [EMAIL PROTECTED] o a [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Systems Admin; Telecom Newbie - What do I need?
On Jul 14, 2005, at 11:54 AM, Adam Goryachev wrote: Use a phone like the polycom IP301/501/600 which has a built-in 2 port 10/100 switch. ie, take the existing cable and plug it into the phone, then take a second cable, connect one end to the phone, and the other to your PC. No need for any additional major investment Independent of my telephony overhaul, I am planning on migrating my network to gigabit to speed up some core file services (we do a lot of server-based computing). Are there phones with a gigabit switch in them? :) Or, kludgy though it seems to me, is it realistic to suggest buying an el-cheapo unmanaged gigabit switch for every office? Looks like I could get away with $50 per node... http://www.cdw.com/shop/products/default.aspx?EDC=652855 From a network admin's perspective, that seems to me like asking for trouble. But routing packets really isn't my specialty, so I don't know if it would really cause any problems. Well, there might be some way you could add a PRI connection from asterisk to your comdial system, then configure your comdial system so that all 'special' service numbers, and external calls are routed to asterisk... Then there is probably a big battle to try and configure everything to forward to the right system to support all the features you want from asterisk etc.. (ie, there must be a reason you are thinking of replacing the existing system?). Yeah, our ComDial DXP is 1980s technology, has been discontinued for years, and most service techs won't even touch it (let alone be able to find parts for it). The idea is to get rid of it, not keep it lingering. But the phones themselves sure are nice ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Festival questions
On Wed, 2005-07-13 at 23:34 -0700, Jason Walker wrote: Has anyone had any luck in changing the voices for Festival and Asterisk? I have Festival installed and working, but can not get the voice different from the default. Thanks, Jason Well, it's been a while since I had to do this so I'm going from memory, but I think you can change the default voice to a different one (assuming you have the voices installed correctly) from the voices.scm file. Look for default-voice-priority-list in that file. The first one in the list is the default. -Seth ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Monitor command stop on call transfer
When i transfer a call to other extension monitor stop recording ZAP1(from out side) - Monitor-Virtual recepcionist-SIP1-attended transfer to SIP2 and monitor crash(stop recording) __ how i can fix it to monitor all the customer call. I have [EMAIL PROTECTED] whit CVS HEAD.-- David Romero## ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SMS on my own possible?
Asterisk + Kannel. When you need to send a message, call the kannel directive with a System call. This is assuming you can connect to a SMSC via SMPP. Jesus Mogollon2005/7/13, Ronald_Wiplinger [EMAIL PROTECTED]: I am thinking of SMS and wonder if I can set-up with Asterisk a SMSC anduse SMS to / from VoIP phones.Can anybody give me a hint? Or has anybody done that?byeRonald___ Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: SpanDSP rxfax, no tiff
Yes, the permissions are okay for getting to that folder.
/var/spool/asterisk is writable (voicemail works that's a subdirectory
under the same path that has the same permissions as the subdirectory
'asterisk-fax'
As the same user that runs asterisk I did a 'touch
/var/spool/asterisk/asterisk-fax/test.tif' just to be sure I could write
to that directory. Permissions are fine.
---
If I leave the mailfax step out entirely, then there should be a .tif
file, right? But theres not. No tif file gets created at all.
Permissions on the fax folder are 777 at the moment.
Are the permissions okay for getting TO that folder? (do you have r+x on
the directories 'above' the fax folder?)
Thanks for the responses so far.
Just trying to help out, I'm using ISDN lines with a DID, for receiving
I do this..
If one calls number X, then Goto(custom-fax,s,1) and then..
[custom-fax]
exten = s,1,Answer
exten = s,2,Macro(faxreceive)
exten = s,3,SetVar(ONZENID=${UNIQUEID})
exten = h,1,system(/usr/local/sbin/mailfax ${FAXFILE}
[EMAIL PROTECTED] ${CALLERIDNUM} ${CALLERIDNAME} ${ONZENID})
[macro-faxreceive]
exten = s,1,SetVar(FAXFILE=/var/spool/asterisk/fax/${UNIQUEID}.tif)
exten = s,2,SetVar(EMAILADDR=${FAX_RX_EMAIL})
exten = s,3,rxfax(${FAXFILE})
exten = s,103,SetVar(EMAILADDR=${FAX_RX_EMAIL})
exten = s,104,Goto(3)
The [macro-faxrecevive] is from AMP (wich I'm using for managing asterisk)
Hope you can do something with this.. :-)
Cheers
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RE: [Asterisk-Users] Systems Admin; Telecom Newbie - What do I ne ed?
Let me expand on the bandwidth point HTH made and maybe shed light on your requirements A 100baseT switched (no hubs) network has a lot of bandwidth when you think in terms of VoIP. The uLaw stream (uncompressed) from an IP500 phone to the Asterisk box is not going to take more than 80K of bandwith from the bandwidth pool. That means 60 phones ALL in a single call would only be using around 5 megs of throughput. At that point packet scheduling becomes far more important than bandwidth. Gigabit is nice but the value of QoS in comparison is very evident. If cost becomes a driving factor, you may want to focus on upgrading port count and remain at 100baseT instead of going to Gigabit. A properly configured 100baseT network with good QoS rules will yield great performance over an unregulated 100baseT network. Do you know your real traffic needs? I would check how much traffic is via user download, www browsing, streaming, email, etc, etc... You may find that some simple rules save you quite a bit of cash. Just a thought and alternative... Gigabit is also very tempting so that whole spiel may have been for not. 8) Also, pay heed to the PoE stuff you are hearing about. I may be wrong but I am pretty sure you want to be careful what you connect to a PoE port. Otherwise you wind up with fried PoE injectors and end devices. I believe PoE ports would only be used for a PoE phone in essence. Just as a reminder and warning. Cheers, Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Colin Anderson Sent: Thursday, July 14, 2005 8:47 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Systems Admin; Telecom Newbie - What do I ne ed? But currently, I only have one ethernet jack per office. Routing another 60 or so ports would add a very substantial expense in both cabling and backbone expansion (what category ethernet is required, BTW?). Most decent phones have an ethernet passthrough (2 port) so you can plug in your PC. As long as your LAN is decent (Cat5 100baseT switched) the overhead using VoIP is negligible. I have used the 3Com NJ wall jacks with good success: http://www.3com.com/products/en_US/detail.jsp?tab=featurespathtype=purc hase sku=3CNJ90 It's basically a 4 port switch that you replace your wall jack with. I used the NJ200, it allows you to set priority per port, although I think they are discontinued now. In combination with a 3Com power over Ethernet injector, I was able to expand a 24 port LAN to a 96 port LAN with a per-port cost of $62 Cdn. And, 24 ports of those 96 are PoE, so I can plug my phones right in to port 1 and they power up, no external power supply needed. hth ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
