Re: [Asterisk-Users] Dialplane slip
Altus Snyman wrote: Good day all My extensions.conf is something like this [main] ;---incoming+ play welcome message extens = s.. ;---users extensions exten = 100. ;---outgoing ignore 0 ;- It all works fine The message says dial 1 for this ens But if I dial 0+number it will actually make a outgoing call! How do I stop this? I must allow the ignore 0 for internal uses but not if a call comes in from the outside? You shouldn't have outgoing in the same context. Reorganise as follows: [incoming] include = extensions include = mainmenu [mainmenu] exten = s... [extensions] exten = 100... [outgoing] ignorepat = 0 ... [internal] include = extensions include = outgoing Then you put context=internal for any phones that are internal (I.E. SIP/FXS/IAX etc), context=incoming for any FXO lines/trunks... -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ 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] FXO and groups
Hi. I have just added two FXO cards in my PC: - Zap/1 is my France Telecom telephone line - Zap/2 is my Free telephone line (Free is an ADSL provider which provides an additional line using VOIP, but this line is only accessible as an FXS, no way to use it directly in H.323 or SIP) In incoming mode, those two lines ring different extensions in my phone installation. In outgoing mode, depending on the dialed number, I'd like to use either one of the lines or both of them; also, depending on the dialed number, either Zap/1 or Zap/2 must be selected in priority. I have tried the following in my zapata.conf: [today's CVS asterisk] [...] context=ftincoming channel = 1 context=freeincoming channel = 2 group=1 channel = 1,2 group=2 channel = 2,1 The idea is to use one of: - Zap/1: France Telecom - Zap/2: Free - Zap/g1: France Telecom if available, Free otherwise - Zap/g2: Free if available, France Telecom otherwise However, when I do that, I cannot use Zap/g1 or Zap/g2. For example, 15 should use Zap/g1, and I get: *CLI dial 15 -- Executing Dial(ALSA/hw:1,0, Zap/g1/15|30) in new stack Jan 24 22:49:07 NOTICE[24747]: app_dial.c:884 dial_exec_full: Unable to create channel of type 'Zap' (cause 0) == Everyone is busy/congested at this time (1:0/0/1) On the other hand, when I try something which should use Zap/g2, I get: *CLI dial 0145815972 -- Executing Dial(ALSA/hw:1,0, Zap/g2/0145815972|30) in new stack -- Called g2/0145815972 *CLI hangup -- Hungup 'Zap/1-1' == Spawn extension (local, 0145815972, 1) exited non-zero on 'ALSA/hw:1,0' which shows that Zap/1 was used while Zap/2 was available and should have been preferred. Is there a way to achieve what I am trying to do? Sam -- Samuel Tardieu -- [EMAIL PROTECTED] -- http://www.rfc1149.net/sam ___ 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] Turn off DTMF recognition pending on CallerID
Is it possible to turn off DTMF recognition (and all transfer services etc.) pending on CallerID (or FXS channel)? Some of the FXS channels I will setup soon, is going to work exactly like POTS. It will be used by people not knowing their within Asterisk. They will be pretty confused when Transfer is playbacked in the handset. :) ___ 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: [Fwd: Re: [Asterisk-biz] bellster.net - GREAT advance]
Duane == Duane [EMAIL PROTECTED] writes: Duane It costs me between 20 and 30c per call to make local calls, so Duane this basically only leaves North American and New Zealand as Duane the only viable options that I know of. In France, the second most important ADSL provider (named Free) offers a phone line (which uses VoIP but can only be used as a FXS) with unlimited free calls to landlines. Sam -- Samuel Tardieu -- [EMAIL PROTECTED] -- http://www.rfc1149.net/sam ___ 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 calls back after phone call
I get the same thing. Its as if the grandstream does'nt send a hangup signal. Someone out there must have fixed this??? Doug -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Kim Lux Sent: Tuesday, January 25, 2005 8:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk calls back after phone call I'm connecting to a commercial SIP provider (ie no * ... yet) and I get the same thing, including the 487. This phone has version x.18 firmware. On Tue, 2005-01-11 at 12:16 +1300, James Doherty wrote: When I call someone, if the call isn't answered and then I hang up, I get 487 coming up on the grandstream phone. If I pick up the receiver again and then hang up, the PBX starts calling me back and when I pickup and listen, there is silence. -- Kim Lux, Diesel Research Inc. ___ 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] :: Success Case = Motorola 62802-51 as FXO device::
Hello, could You spare some more details about this? Any source code modifications? Greetings, Pawel - Original Message - From: Jefferson Carvalho [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, January 19, 2005 2:41 PM Subject: [Asterisk-Users] :: Success Case = Motorola 62802-51 as FXO device:: Hello list , I´d like to report a success case with a modem based on chipset : Motorola 62802-51. It works fine , and zaptel identifies as a X100P ( not clone ) . Red Alarms can be identified . :) This doesn´t occurred on MD3200 ambient chipsets. Best Regards , -- - Jefferson Carvalho Analista de Redes / Com. de dados Credishop S/A Voip: sip:[EMAIL PROTECTED] DDR : 55.86.2106-1243 Móvel: 55.86.9432-1901 ___ 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] IP FXS channel bank
el Flynn wrote: Hi there, I'm trying to get hold of an evaluation IP-enabled FXS channel bank. I'm not sure if it's more a channel bank, or should be called a multiport-ATA. Oh well. On the surface it looks quite nice - 16 FXS ports, and since it's connected to the network I can do away with an E1/T1 connection required of the normal channel banks (if it can be called that :) Here are some features I got from the brochure: 1. MGCP, H.323 (v4) and SIP support 2. Selectable, multiple codes (g711/g723/g729A) per channel 3. G.168/165-compliant adaptive echo cancellation 4. Echo canceller jitter buffer, VAD and CNG 5. complete voice band signalling support 6. provides inband/outband DTMF generation/detection 7. provides call progress tones 8. web management interface 9. LAN (10/100) and WAN RJ-45 ports It looks like a standart VoIP box, not like a channel bank. Of course it will be worse than a T1 card with CAC Channel bank. ___ 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] Bristuff ZapHFC and Loosing D-Channel
Using the latest(?) bristuff (Asterisk 1.0.4-BRIstuffed-0.2.0-RC3a) I have problems with loosing the D-channel. Most of the time, after the message PRI D-channel down it only takes a second or so to come back up, noted by the message PRI D-channel up However, today most of the time the D-channel stays down. Calls come in, but can't be answered. Does anyone know of a fix for this, or might have some insights on how to circumvent this problem? Any and all help is greatly appreciated. -- Best regards Peer Oliver Schmidt ___ 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] SER Prob
Hi all,
Hope somebody can help-I really am stumped as to why this won't work.
I realise that this isnt an Asterisk problem (Please dont bash me on
the list) and I have emailed the SER list but I havent received a
reply and maybe someone on this list can help...Once this problem is
solved I am going to use Asterisk for voicemail etc with SER (I have
it set up)
I currently have SER set up and clients are registering successfully.
However I want clients to authenticate before they can register.
Howevere when I uncomment the relevant lines in the ser.cfg file, my
clients can't register. The only thing I can think of is that SER is
behind NAT and my clients may/may not be behind NATI have
included my ser.cfg file below...I have spent along time trying to
understand why this is happening so any help will be appreciated!
Thanks,
Aisling.
#
# $Id: ser.cfg,v 1.21.4.1 2003/11/10 15:35:15 andrei Exp $
#
# simple quick-start config script
#
# --- global configuration parameters
#debug=3 # debug level (cmd line: -dd)
#fork=yes
#log_stderror=no # (cmd line: -E)
/* Uncomment these lines to enter debugging mode
debug=7
fork=no
log_stderror=yes
*/
check_via=no # (cmd. line: -v)
dns=no # (cmd. line: -r)
rev_dns=no # (cmd. line: -R)
port=5060
#children=4
fifo=/tmp/ser_fifo
alias=84.203.148.14
# -- module loading --
# Uncomment this if you want to use SQL database
#loadmodule /usr/lib/ser/modules/mysql.so
loadmodule /usr/lib/ser/modules/sl.so
loadmodule /usr/lib/ser/modules/tm.so
loadmodule /usr/lib/ser/modules/rr.so
loadmodule /usr/lib/ser/modules/maxfwd.so
loadmodule /usr/lib/ser/modules/usrloc.so
loadmodule /usr/lib/ser/modules/registrar.so
loadmodule /usr/lib/ser/modules/nathelper.so
#loadmodule /usr/lib/ser/modules/mediaproxy.so
loadmodule /usr/lib/ser/modules/textops.so
#loadmodule /usr/lib/ser/modules/maxfwd.so
# Uncomment this if you want digest authentication
# mysql.so must be loaded !
#loadmodule /usr/lib/ser/modules/auth.so
#loadmodule /usr/lib/ser/modules/auth_db.so
# - setting module-specific parameters ---
# -- usrloc params --
modparam(usrloc, db_mode, 0)
# Uncomment this if you want to use SQL database
# for persistent storage and comment the previous line
#modparam(usrloc, db_mode, 2)
# -- auth params --
# Uncomment if you are using auth module
#
#modparam(auth_db, calculate_ha1, yes)
#
# If you set calculate_ha1 parameter to yes (which true in this
config),
# uncomment also the following parameter)
#
#modparam(auth_db, password_column, password)
# -- rr params --
# add value to ;lr param to make some broken UAs happy
modparam(rr, enable_full_lr, 1)
#!!Nathelper
#modparam(registrar,nat_flag,6)
#modparam(nathelper,natping_interval,30) #Ping intervals 30
seconds
#modparam(nathelper,ping_nated_only,1) #Ping only clinets
behind NAT
# -request routing logic---
# main routing logic
route{
# initial sanity checks -- messages with
# max_forwards==0, or excessively long requests
if (!mf_process_maxfwd_header(10)) {
sl_send_reply(483,Too Many Hops);
break;
};
if ( msg:len max_len ) {
sl_send_reply(513, Message too big);
break;
};
#Aisling Insert
# #!Nat Insert
# #the below line tests if the IP of the received packet is different
from the IP in the via header and also
# #sees if the IP address in the contact header is private
# if (nat_uac_test(3)){
# if (method == REGISTER || ! search(^Record-Route:)){
# log(Log: Someone trying to register from private
IP,rewriting\n);
# # fixed_nated_contact(); #Rewrite contact with source IP
# if (method == INVITE){
# fix_nated_sdp(1); #Add direction=active to SDP
# };
# force_rport(); # Add rport parameter to topmost Via
# setflag(6); # Mark as Nated
# };
# };
###End#
# we record-route all messages -- to make sure that
# subsequent messages will go through our proxy; that's
# particularly good if upstream and downstream entities
# use different transport protocol
if (!method == REGISTER) record_route();
# loose-route processing
if (loose_route()) {
t_relay();
break;
};
# if the request is for other domain use UsrLoc
# (in case, it does not work, use the following command
# with proper names and addresses in it)
if (uri==myself) {
if (method==REGISTER) {
# Uncomment this if you want to use digest authentication
# if (!www_authorize(84.203.148.14, subscriber)) {
# www_challenge(84.203.148.14, 0);
# break;
# };
save(location);
break;
};
# native SIP destinations are handled using our USRLOC DB
if (!lookup(location)) {
sl_send_reply(404, Not Found);
break;
};
};
#inserted by klaus
if (method == INVITE){
record_route();
force_rtp_proxy();
/* set up reply processing*/
t_on_reply(1);
};
# forward to current uri now; use stateful forwarding; that
# works reliably even if we forward from TCP to UDP
if (!t_relay()) {
sl_reply_error();
Re: [Asterisk-Users] AVM Fritz crash
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Tue, 25 Jan 2005, Thomas Niesel wrote: Does anyone have any suggestions? Quick shot: SMP, HT? It is an SMP machine but it's only running a uniprocessor kernel ATM (I tried the driver in SMP mode and it crashed in a completely different way so I thought it might be better to debug the problem in UP mode which would hopefully be slightly more tested - in SMP mode the module modprobes ok but if you cat /proc/capi/controllers/1 then it immediately oopses). IRQ? Try other slot. I've tried several slots with the same results. Do you modprobe/insmod the module by hand or via capiinit start? Modprobed by hand. I'm currently running the Fedora Core 3 2.6.10-1.741_FC3 kernel and it's on Athlon hardware. - Steve Jabber: [EMAIL PROTECTED] Web: http://www.nexusuk.org/ Servatis a periculum, servatis a maleficum - Whisper, Evanescence -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.6 (GNU/Linux) Comment: Public key available at http://www.nexusuk.org/pubkey.txt iD8DBQFB9hYS5zUOsIV3bqERAokeAJ9zMNa3j6umz0/dkxzD4BbeqzaRNQCfdJnI QuoEpGtwYZHD+1vo2SVSA5I= =pZZR -END PGP 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
Re: [Asterisk-Users] OT: pinout for standard telephone headsetrequired.?
Hi, this is the pinout of the handset jack of the ciso phone, which again is different to the headset pinout of the cisco. If I had an old pots headset, I could smash^h^h^h^h dismantle it and trace the wires to mic and earpiece but I dont have one I can take apart. Cheers Mike On Tue, 25 Jan 2005 08:31:50 +0100, Florian Overkamp [EMAIL PROTECTED] wrote: Hi, -Original Message- Thanks but no, this is the info I have. I need the pinout for a STANDARD telephone headset. http://www.mml.uni-hannover.de/einhorn/headset/index_e.html On the bottom of the page there is a section 'Notes' that includes pinout for the handset jack. I think that connection is pretty much the standard, although I never tried it :) Florian ___ 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] BUSY-tone on incoming calls?
Is it possible to make the telco send an busy signal when an incoming call are supposed to dial a group which has all lines busy? Since I will get many public phonenumbers into my E1 (from telco), it will be sliced up into a few groups. There might be channels availible in the E1, but not on the other side of Asterisk (the office side). ___ 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] OT: pinout for standard telephoneheadsetrequired.?
Hi, -Original Message- this is the pinout of the handset jack of the ciso phone, which again is different to the headset pinout of the cisco. If I had an old pots headset, I could smash^h^h^h^h dismantle it and trace the wires to mic and earpiece but I dont have one I can take apart. Yes, and I just unplugged handsets from three different non-cisco phones and plugged them into the handset jack of a 7960, it works. You may consider the pin layout from that webpage to be 'standard'. Florian ___ 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] BUSY-tone on incoming calls?
Hi, -Original Message- Is it possible to make the telco send an busy signal when an incoming call are supposed to dial a group which has all lines busy? Since I will get many public phonenumbers into my E1 (from telco), it will be sliced up into a few groups. There might be channels availible in the E1, but not on the other side of Asterisk (the office side). You can set a PRI_CAUSE variable. See http://www.voip-info.org/tiki-index.php?page=Asterisk%20variable%20PRI_CAUSE Florian ___ 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] OT: pinout for standard telephoneheadsetrequired.?
We found that the plantronics headet that we used to have for the Meridian phones did not work in the cisco headset jack. We had to cut the ends off and rewire them, which we worked out by trial and error. You can buy plantronics for cisco phones as well. Julian. Florian Overkamp wrote: Hi, -Original Message- this is the pinout of the handset jack of the ciso phone, which again is different to the headset pinout of the cisco. If I had an old pots headset, I could smash^h^h^h^h dismantle it and trace the wires to mic and earpiece but I dont have one I can take apart. Yes, and I just unplugged handsets from three different non-cisco phones and plugged them into the handset jack of a 7960, it works. You may consider the pin layout from that webpage to be 'standard'. Florian ___ 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] OT: pinout forstandard telephoneheadsetrequired.?
Hi, -Original Message- We found that the plantronics headet that we used to have for the Meridian phones did not work in the cisco headset jack. We had to cut the ends off and rewire them, which we worked out by trial and error. You can buy plantronics for cisco phones as well. Actually, we've worked with plantronics sets and they came with two wires, one that matches the handset plug of the cisco, and one that matches the headset plug of the cisco... YMMV I guess. Florian ___ 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] OT: pinout forstandard telephoneheadsetrequired.?
Damn, that would have saved us a lot of time. We recently had to modify 55 headsets ... :( Julian. Florian Overkamp wrote: Hi, -Original Message- We found that the plantronics headet that we used to have for the Meridian phones did not work in the cisco headset jack. We had to cut the ends off and rewire them, which we worked out by trial and error. You can buy plantronics for cisco phones as well. Actually, we've worked with plantronics sets and they came with two wires, one that matches the handset plug of the cisco, and one that matches the headset plug of the cisco... YMMV I guess. Florian ___ 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] Re: Zapata in Australia
Hi Howard, could you share your indications.conf settings as well? I appreaciate that. manny Howard wrote: This works for me in AU.In /etc/zaptel.conf:fxsks=1loadzone = audefaultzone=auIn /etc/asterisk/zapata.conf:[channels]context = defaultsignalling = fxs_ksechocancel = 128echocancelwhenbridged = yesechotraining = yesrelaxdtmf = yespulsedial = yesrxgain = +15%txgain = +5%immediate = nobusydetect = yesbusycount = 3callprogress = yesmusiconhold = defaultusecallerid = yescallerid = asreceiveduseincomingcalleridonzaptransfer = yesfaxdetect = bothgroup = 1channel = 1Note that I do not get callerid but I do get fax. ___ 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: Zapata in Australia
Emanuele Venditti wrote: Hi Howard, could you share your indications.conf settings as well? I appreaciate that. manny Correct indications for Australia was merged into the CVS long ago... -- Best regards, Duane http://www.cacert.org - Free Security Certificates http://www.nodedb.com - Think globally, network locally http://www.sydneywireless.com - Telecommunications Freedom http://happysnapper.com.au - Sell your photos over the net! http://e164.org - Using Enum.164 to interconnect asterisk servers I do not try to dance better than anyone else. I only try to dance better than 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] Definity PBX with a T100P TN767E
Ken Godee wrote: We have been able to get our Definity G3R working with Asterisk via a T100P card and a TN767E card, works very well! But, I'm a little stuck on how to get the DID info from the G3 and ext/ext info to the G3. Incoming shows the trunk info setup by our phone admin. With no trail(time really) to follow I've given up on trying to get CID info to work properly. Ken, We'll continue to work on it, I'll post the info if we get results. Thanks again, Doug ___ 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] Avoided deadlock
Can anyone shed any light on why I am getting so many warnings on this particular channel - the lady using the phone says that she hasn't had any problems today. However, the number of warnings is concerning to me. As you can see, the vast majority of the issues seem to be with the one phone. Julian. Jan 25 11:06:11 WARNING[24146]: Avoided deadlock for 'SIP/6711-89c7', 10 retries! Jan 25 11:06:20 WARNING[24146]: Avoided deadlock for 'SIP/6711-89c7', 10 retries! Jan 25 11:06:20 WARNING[24146]: Avoided deadlock for 'SIP/6711-89c7', 10 retries! Jan 25 11:06:25 WARNING[24146]: Avoided deadlock for 'SIP/6711-89c7', 10 retries! Jan 25 11:06:29 WARNING[24146]: Avoided deadlock for 'SIP/6711-89c7', 10 retries! Jan 25 11:06:29 WARNING[24146]: Avoided deadlock for 'SIP/6711-89c7', 10 retries! Jan 25 11:07:01 WARNING[24146]: Avoided deadlock for 'SIP/6711-89c7', 10 retries! Jan 25 11:07:02 WARNING[24146]: Avoided deadlock for 'SIP/6711-89c7', 10 retries! Jan 25 11:07:03 WARNING[24146]: Avoided deadlock for 'SIP/6711-89c7', 10 retries! Jan 25 11:07:04 WARNING[24146]: Avoided deadlock for 'SIP/6711-89c7', 10 retries! Jan 25 11:07:04 WARNING[24146]: Avoided deadlock for 'SIP/6711-89c7', 10 retries! Jan 25 11:07:04 WARNING[24146]: Avoided initial deadlock for 'Agent/6017', 10 retries! Jan 25 11:07:04 WARNING[24146]: Avoided deadlock for 'SIP/6711-89c7', 10 retries! Jan 25 11:07:05 WARNING[24146]: Avoided deadlock for 'SIP/6711-89c7', 10 retries! Jan 25 11:07:05 WARNING[24146]: Avoided deadlock for 'SIP/6711-89c7', 10 retries! Jan 25 11:07:05 WARNING[24146]: Avoided deadlock for 'SIP/6711-89c7', 10 retries! Jan 25 11:07:06 WARNING[24146]: Avoided deadlock for 'SIP/6711-89c7', 10 retries! Jan 25 11:07:10 WARNING[24146]: Avoided deadlock for 'SIP/6711-89c7', 10 retries! Jan 25 11:07:17 WARNING[24146]: Avoided deadlock for 'SIP/6711-89c7', 10 retries! Jan 25 11:07:17 WARNING[24146]: Avoided deadlock for 'SIP/6711-89c7', 10 retries! Jan 25 11:07:17 WARNING[24146]: Avoided deadlock for 'SIP/6711-89c7', 10 retries! Jan 25 11:07:23 WARNING[24146]: Avoided deadlock for 'SIP/6711-89c7', 10 retries! Jan 25 11:07:23 WARNING[24146]: Avoided deadlock for 'SIP/6711-89c7', 10 retries! Jan 25 11:07:27 WARNING[24146]: Avoided deadlock for 'SIP/6711-89c7', 10 retries! Jan 25 11:07:31 WARNING[24146]: Avoided deadlock for 'SIP/6711-89c7', 10 retries! Jan 25 11:07:31 WARNING[24146]: Avoided deadlock for 'SIP/6711-89c7', 10 retries! Jan 25 11:07:31 WARNING[24146]: Avoided deadlock for 'SIP/6711-89c7', 10 retries! Jan 25 11:07:35 WARNING[24146]: Avoided deadlock for 'SIP/6711-89c7', 10 retries! Jan 25 11:07:37 WARNING[24146]: Avoided deadlock for 'SIP/6711-89c7', 10 retries! Jan 25 11:07:37 WARNING[24146]: Avoided deadlock for 'SIP/6711-89c7', 10 retries! Jan 25 11:07:37 WARNING[24146]: Avoided deadlock for 'SIP/6711-89c7', 10 retries! Jan 25 11:07:37 WARNING[24146]: Avoided deadlock for 'SIP/6711-89c7', 10 retries! Jan 25 11:07:39 WARNING[24146]: Avoided deadlock for 'SIP/6711-89c7', 10 retries! Jan 25 11:07:40 WARNING[24146]: Avoided deadlock for 'SIP/6711-89c7', 10 retries! Jan 25 11:07:40 WARNING[24146]: Avoided deadlock for 'SIP/6711-89c7', 10 retries! Jan 25 11:07:46 WARNING[24146]: Avoided deadlock for 'SIP/6711-89c7', 10 retries! Jan 25 11:07:46 WARNING[24146]: Avoided deadlock for 'SIP/6711-89c7', 10 retries! Jan 25 11:07:49 WARNING[24146]: Avoided deadlock for 'SIP/6711-89c7', 10 retries! Jan 25 11:07:49 WARNING[24146]: Avoided deadlock for 'SIP/6711-89c7', 10 retries! Jan 25 11:07:51 WARNING[24146]: Unable to forward frame Jan 25 11:07:51 WARNING[24146]: Avoided deadlock for 'SIP/6711-89c7', 10 retries! Jan 25 11:07:51 WARNING[24146]: Avoided deadlock for 'SIP/6711-89c7', 10 retries! Jan 25 11:07:58 WARNING[24146]: Avoided deadlock for 'SIP/6711-89c7', 10 retries! Jan 25 11:07:59 WARNING[24146]: Avoided deadlock for 'SIP/6711-89c7', 10 retries! Jan 25 11:08:03 WARNING[24146]: Avoided deadlock for 'SIP/6711-89c7', 10 retries! Jan 25 11:08:14 WARNING[24146]: Avoided deadlock for 'SIP/6711-89c7', 10 retries! Jan 25 11:08:21 WARNING[24146]: Avoided deadlock for 'SIP/6711-89c7', 10 retries! Jan 25 11:08:21 WARNING[24146]: Avoided deadlock for 'SIP/6711-89c7', 10 retries! ___ 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] Mediatrix voip gateway 1124 and 1204 in UKsetting
Many thanks for that info! Peter One thing to consider if you only have 3 PSTN lines is the Sipura SPA-3000 (you would need 3 of them, one for each line) We have 2 PSTN lines at our scout campsite, and they work very well, as well as providing a simple power outage solution. They retail about £80 + the VAT I can supply more information once you have looked at the devices. Regards David On 24 Jan 2005, at 11:20, Peter Hoppe wrote: Hello! We are located in the UK, and we are planning to replace our old pbx with an asterisk based pbx. For outgoing calls our present pbx is connected to three PSTN lines which all have the same number. Internally, the pbx caters for quite a few extensions, and each extension can make outbound phone calls. Only very rarely does our call volume exceed three simultaneous connections (inside to inside plus inside to outside). We have looked into the issue of connecting the phones and the outside lines to the system. For the fxo connectivity we want to stick with the three PSTN lines, because they worked for us and we don't see a need to upgrade to ISDN. The asterisk system will be also connected to the internet anyway so we can perform VOIP calls. For the fxs connectivity we want to re-use the old telephone wiring and provide standard two-wire telephones. Putting in IP phones would mean a massive installation effort, as we would have to put an entire new computer network in place - plus many IP phones constantly connected to mains, plus admin headaches, plus security issues and so on. The two wire solution seems the best solution for our setting. We have looked into using a channel bank for the analog conectivity, and we are currently in contact with Carrier Access to purchase a new Adit 600 unit with space for 48 extensions. We cannot provide fxo connectivity via the channel bank because the fxo card from CA seems not to be EU approved. One downside of the channel bank is that we need a special T1 card for it to operate with the asterisk pbx. Also, channel banks seems to be a particular US concept, so we would have difficulties to get replacement parts, if something breaks. Recently I heard of the alternative solution of a voip gateway, and the particular units I have seen are the Mediatrix 1124 for fxs connection and the Mediatrix 1204 for the fxo connection. Both units support the SIP protocol, so it should be possible to connect them to the asterisk PC via standard network connection. Mediatrix seems to have resellers in Europe as well, so it might be possible that their devices are Europe approved as well. Question: * Does anyone have any experience with these units in a UK setting? * For the 1124: Does it work with standard UK two wire phones? Are there impedance problems (especially concerning echo problems)? Is the audio quality sufficient? Are they transparent to the asterisk system, i.e. does each fxs port look like a separate IP phone to the asterisk system? * For the 1204: Would it be approved for connection into the UK PSTN (The prospectus from Mediatrix didn't say anything about regulatory approvals)? Can they initiate outside calls / receive incoming calls or are there problems (signalling compatible with UK PSTN)? Are they transparent to the asterisk system, i.e.does each fxo port look like a separate IP phone to the asterisk system? I do realize that these questions are quite broad, but do appreciate any info. Thank you very much for your consideration. ___ 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 subject)
Pat Delaney wrote: Thanks for you comments. I have the one port card now. I plan on purchasing the TDM400. My only question is whether or not the Dell optiplex has pci 2.1 (I think) Depends on the model, check Dells website. A quick googling show: Specifications: *OptiPlex* GXi. *...* System/video BIOS chip, 1 Mbit (128 KB). BIOS core, *Dell* Phoenix. *...* *PCI* bus specification, complies with *PCI* specification 2.1. *...* Doug ___ 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] Problems with H323 channels
Hello, I trying to set up an h323 channel over TCP/IP network to connect two PBX. I just read http://www.voip-info.org/wiki-Asterisk+config+h323.conf but, it don't solve my dubs. How could I use a h323 channel with asterisk? Could anyone paste a part of h323.conf file? I am no sure how to setting up h323.conf. And the part of extensions.conf where you use the h323 channels for an specific prefix? Thanks. Ismael. ___ 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] OT: pinout forstandard telephoneheadsetrequired.?
Julian, can you remember how you wired it, ie which pin on the plantronics connector went to which pin on the ciso? Mike On Tue, 25 Jan 2005 10:32:50 +, Asterisk [EMAIL PROTECTED] wrote: Damn, that would have saved us a lot of time. We recently had to modify 55 headsets ... :( Julian. Florian Overkamp wrote: Hi, -Original Message- We found that the plantronics headet that we used to have for the Meridian phones did not work in the cisco headset jack. We had to cut the ends off and rewire them, which we worked out by trial and error. You can buy plantronics for cisco phones as well. Actually, we've worked with plantronics sets and they came with two wires, one that matches the handset plug of the cisco, and one that matches the headset plug of the cisco... YMMV I guess. Florian ___ 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] OT: pinout forstandard telephoneheadsetrequired.?
I'm going into work now, and will send the specs when I'm there. Julian. Mike Dent wrote: Julian, can you remember how you wired it, ie which pin on the plantronics connector went to which pin on the ciso? Mike On Tue, 25 Jan 2005 10:32:50 +, Asterisk [EMAIL PROTECTED] wrote: Damn, that would have saved us a lot of time. We recently had to modify 55 headsets ... :( Julian. Florian Overkamp wrote: Hi, -Original Message- We found that the plantronics headet that we used to have for the Meridian phones did not work in the cisco headset jack. We had to cut the ends off and rewire them, which we worked out by trial and error. You can buy plantronics for cisco phones as well. Actually, we've worked with plantronics sets and they came with two wires, one that matches the handset plug of the cisco, and one that matches the headset plug of the cisco... YMMV I guess. Florian ___ 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 ___ 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] Turn off DTMF recognition pending on CallerID
Daniel Nyström wrote: Is it possible to turn off DTMF recognition (and all transfer services etc.) pending on CallerID (or FXS channel)? Some of the FXS channels I will setup soon, is going to work exactly like POTS. It will be used by people not knowing their within Asterisk. They will be pretty confused when Transfer is playbacked in the handset. :) Don't enable the feature then. ___ 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] BUSY-tone on incoming calls?
Hi, -Original Message- You can set a PRI_CAUSE variable. See http://www.voip-info.org/tiki-index.php?page=Asterisk%20variab le%20PRI_CAUSE This only works in CVS-HEAD. For production use just run Busy() in the dialplan. Actually, we use this on 1.0.3 with BRI-STUFF. Florian ___ 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] Correct way to update Asterisk
Thanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of steve szmidt Sent: Monday, January 24, 2005 11:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Correct way to update Asterisk On Monday 24 January 2005 23:31, Pat Delaney wrote: Pardon the newbie post. I installed Asterisk on a test system using the [EMAIL PROTECTED] cd image. When you boot from [EMAIL PROTECTED] is installs an O/S and Asterisk on your PC. How cool is that. But I was wondering if someone could point me in the right direction for updating the version that I have. I'm new to CVS, how do I determine what version to build? Is there a primer on how to download the latest version and install it? If I manage to figure out how to pull it down, when I build it and install, will it overwrite my configurations? Sorry again for the dumb questions Pat You could try my update script at: szmidt.org/asterisk/asterisk-update.sh It will backup what you have and update, compile and install it for you. You can even do it in a number of different ways. -- Steve Szmidt They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ 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] SER Prob
try something like this:
# Uncomment this if you want to use digest authentication
if (!www_authorize(, subscriber)) {
www_challenge(, 0);
break;
};
maybe you have Problems with your realm.
And this seems not to be the list where you can find good help for your
Problem!
Best Regards
markus
Am Die, den 25.01.2005 schrieb Ashling O'Driscoll um 10:42:
Hi all,
Hope somebody can help-I really am stumped as to why this won't work.
I realise that this isnt an Asterisk problem (Please dont bash me on
the list) and I have emailed the SER list but I havent received a
reply and maybe someone on this list can help...Once this problem is
solved I am going to use Asterisk for voicemail etc with SER (I have
it set up)
I currently have SER set up and clients are registering successfully.
However I want clients to authenticate before they can register.
Howevere when I uncomment the relevant lines in the ser.cfg file, my
clients can't register. The only thing I can think of is that SER is
behind NAT and my clients may/may not be behind NATI have
included my ser.cfg file below...I have spent along time trying to
understand why this is happening so any help will be appreciated!
Thanks,
Aisling.
#
# $Id: ser.cfg,v 1.21.4.1 2003/11/10 15:35:15 andrei Exp $
#
# simple quick-start config script
#
# --- global configuration parameters
#debug=3 # debug level (cmd line: -dd)
#fork=yes
#log_stderror=no # (cmd line: -E)
/* Uncomment these lines to enter debugging mode
debug=7
fork=no
log_stderror=yes
*/
check_via=no # (cmd. line: -v)
dns=no # (cmd. line: -r)
rev_dns=no # (cmd. line: -R)
port=5060
#children=4
fifo=/tmp/ser_fifo
alias=84.203.148.14
# -- module loading --
# Uncomment this if you want to use SQL database
#loadmodule /usr/lib/ser/modules/mysql.so
loadmodule /usr/lib/ser/modules/sl.so
loadmodule /usr/lib/ser/modules/tm.so
loadmodule /usr/lib/ser/modules/rr.so
loadmodule /usr/lib/ser/modules/maxfwd.so
loadmodule /usr/lib/ser/modules/usrloc.so
loadmodule /usr/lib/ser/modules/registrar.so
loadmodule /usr/lib/ser/modules/nathelper.so
#loadmodule /usr/lib/ser/modules/mediaproxy.so
loadmodule /usr/lib/ser/modules/textops.so
#loadmodule /usr/lib/ser/modules/maxfwd.so
# Uncomment this if you want digest authentication
# mysql.so must be loaded !
#loadmodule /usr/lib/ser/modules/auth.so
#loadmodule /usr/lib/ser/modules/auth_db.so
# - setting module-specific parameters ---
# -- usrloc params --
modparam(usrloc, db_mode, 0)
# Uncomment this if you want to use SQL database
# for persistent storage and comment the previous line
#modparam(usrloc, db_mode, 2)
# -- auth params --
# Uncomment if you are using auth module
#
#modparam(auth_db, calculate_ha1, yes)
#
# If you set calculate_ha1 parameter to yes (which true in this
config),
# uncomment also the following parameter)
#
#modparam(auth_db, password_column, password)
# -- rr params --
# add value to ;lr param to make some broken UAs happy
modparam(rr, enable_full_lr, 1)
#!!Nathelper
#modparam(registrar,nat_flag,6)
#modparam(nathelper,natping_interval,30) #Ping intervals 30
seconds
#modparam(nathelper,ping_nated_only,1) #Ping only clinets
behind NAT
# -request routing logic---
# main routing logic
route{
# initial sanity checks -- messages with
# max_forwards==0, or excessively long requests
if (!mf_process_maxfwd_header(10)) {
sl_send_reply(483,Too Many Hops);
break;
};
if ( msg:len max_len ) {
sl_send_reply(513, Message too big);
break;
};
#Aisling Insert
# #!Nat Insert
# #the below line tests if the IP of the received packet is different
from the IP in the via header and also
# #sees if the IP address in the contact header is private
# if (nat_uac_test(3)){
# if (method == REGISTER || ! search(^Record-Route:)){
# log(Log: Someone trying to register from private
IP,rewriting\n);
# # fixed_nated_contact(); #Rewrite contact with source IP
# if (method == INVITE){
# fix_nated_sdp(1); #Add direction=active to SDP
# };
# force_rport(); # Add rport parameter to topmost Via
# setflag(6); # Mark as Nated
# };
# };
###End#
# we record-route all messages -- to make sure that
# subsequent messages will go through our proxy; that's
# particularly good if upstream and downstream entities
# use different transport protocol
if (!method == REGISTER) record_route();
# loose-route processing
if (loose_route()) {
t_relay();
break;
};
# if the request is for other domain use UsrLoc
# (in case, it does not work, use the following command
# with proper names and addresses in it)
if (uri==myself) {
if (method==REGISTER) {
# Uncomment this if you want to use digest authentication
# if
Re: [Asterisk-Users] AVM Fritz crash
Hallo Steve Hill On Tue, 25 Jan 2005 09:49:05 + (GMT) you wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Tue, 25 Jan 2005, Thomas Niesel wrote: Does anyone have any suggestions? Quick shot: SMP, HT? It is an SMP machine but it's only running a uniprocessor kernel ATM (I tried the driver in SMP mode and it crashed in a completely different way so I thought it might be better to debug the problem in UP mode which would hopefully be slightly more tested - in SMP mode the module modprobes ok but if you cat /proc/capi/controllers/1 then it immediately oopses). Hm... IRQ? Try other slot. I've tried several slots with the same results. Ok Do you modprobe/insmod the module by hand or via capiinit start? Modprobed by hand. Do you have a valid capi.conf (for the card, not for asterisk) Try using capiinit start. Finally try another card or try the card in another box I'm currently running the Fedora Core 3 2.6.10-1.741_FC3 kernel and it's on Athlon hardware. - Steve Jabber: [EMAIL PROTECTED] Web: http://www.nexusuk.org/ Servatis a periculum, servatis a maleficum - Whisper, Evanescence -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.6 (GNU/Linux) Comment: Public key available at http://www.nexusuk.org/pubkey.txt iD8DBQFB9hYS5zUOsIV3bqERAokeAJ9zMNa3j6umz0/dkxzD4BbeqzaRNQCfdJnI QuoEpGtwYZHD+1vo2SVSA5I= =pZZR -END PGP 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 -- Tho/\/\as ___ 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] OH323 Cisco Transfer Key
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi All I'm using * as a Call Center to CCM. All the phones are ciscop ip phones (witk skinny) attached to CCM. When i try to transfer a call, from one phone to another, when i press the transfer key i get this message on oh323.log: ~ [2]PAsteriskSoundChannel::Write: Write Failed (G.711) - Destination Address Required and i can't transfer the calls because the channels are broken. However i can transfer the call using the # key (via asterisk), but i want to know if is possible to do this using the cisco transfer key (via ccm). Thanks in advance Joo Amaro -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) iD8DBQFB9j/OJUm/Bor63CERAnMYAJ9ww1VHxZ/YP8fIurUTMcFxrp8IoACfbvj/ VwA59Os8h5SLmr67YwMn1wI= =0/WZ -END PGP 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
Re: [Asterisk-Users] Nufone and Dialing Out
On January 24, 2005 11:21 pm, Bobby Lacey wrote: Yes I have it there. Here is my iax.conf [NuFone] type=user secret=pass context=inbound That is a user entry. I'm looking for a peer entry, which is used when placing outbound calls. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Nufone and Dialing Out
Andrew Kohlsmith wrote: On January 24, 2005 11:21 pm, Bobby Lacey wrote: Yes I have it there. Here is my iax.conf [NuFone] type=user secret=pass context=inbound That is a user entry. I'm looking for a peer entry, which is used when placing outbound calls. If he's dialing via Dial(IAX2/username:[EMAIL PROTECTED] then I don't think Asterisk will use the peer entry anyway. ___ 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] BUSY-tone on incoming calls?
On Tue, 25 Jan 2005, Eric Wieling wrote: Florian Overkamp wrote: You can set a PRI_CAUSE variable. See http://www.voip-info.org/tiki-index.php?page=Asterisk%20variable%20PRI_CAUSE This only works in CVS-HEAD. For production use just run Busy() in the dialplan. It was added to Asterisk 2003/11/05, so it should be in _all_ 1.0 releases. 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
RE: [Asterisk-Users] Updating Asterisk
Just tried it. Show version still shows: Connected to Asterisk CVS-v1-0-12/21/04-14:14:46 Which is exactly what it said prior to the upgrade. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Monday, January 24, 2005 10:31 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Updating Asterisk How do I know if the update occurred? After downloading from CVS, I did make clean, make install and then stopped and started Asterisk. If I'm not mistaken, before you do make install you have to stop asterisk, else it can't be replaced because it's used. so, just go in you asterisk source dir, stop asterisk then make install and after that restart it 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 The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ 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] BUSY-tone on incoming calls?
On Tue, 25 Jan 2005, Eric Wieling wrote:
You can set a PRI_CAUSE variable. See
http://www.voip-info.org/tiki-index.php?page=Asterisk%20variable%20PRI_CAUSE
This only works in CVS-HEAD. For production use just run Busy() in the
dialplan.
No, PRI_CAUSE works great at least in 1.0.2, probably in the earlier 1.0
releases too. Busy() may play a busy tone to the caller instead of
signalling busy so using PRI_CAUSE is much better in PRI or BRI
environment.
exten = 123437,1,Dial(Zap/g2/37,26,tg)
exten = 123437,2,GotoIf($[${DIALSTATUS} = BUSY]?110:3)
exten = 123437,3,Answer
exten = 123437,4,Wait(1)
exten = 123437,5,Voicemail(su21)
exten = 123437,6,Hangup
exten = 123437,110,SetVar(PRI_CAUSE=17)
exten = 123437,111,Hangup
--
Regards,
Tobias Jönsson, Lund SE___
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[Asterisk-Users] Codec mismatch between SIP (BT) and IAX Phone
Hi, I have strange problem. I have 1 SIP client (bt100) and 1 Iax2 client (IAXPhone): - when I call from Iax to SIP sound works - when I call from Sip to Iax sound doesn't work, I get : Jan 25 13:52:22 NOTICE[31334]: channel.c:1314 ast_read: Dropping incompatible voice frame on IAX2/200/1 of format gsm since our native format has changed to ulaw Why is Asterisk not satisfied with gsm packets - it should transcode if necessary ? I have enabled gsm and ulaw in both configs, but it seems not sufficient. Any advice, help ? Thanks in advance, regards, Rob. In both configs there are only general codec settings . I have in sip.conf (snippet): [general] port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0; Address to bind to (all addresses on machine) disallow=all allow=ulaw allow=alaw allow=gsm context = from-sip ; Send unknown SIP callers to this context And in iax.conf (snippet) : [general] bindport = 4569 ; Port to bind to (IAX is 4569) bindaddr = 0.0.0.0; Address to bind to (all addresses on machine) ;delayreject=yes disallow=all allow=ulaw allow=alaw allow=gsm jitterbuffer=yes mailboxdetail=yes authdebug=yes ___ 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] LiveVoip DTMF Issues
My assumption is that most folks trunking through Voicepulse Connect must be using SIP since I haven't seen this problem mentioned before. So my conclusion is that DTMF and SIP and VPC work fine together BUT then you don't get to benefit from the efficiency of IAX. So the million dollar question is: Does IAX have a problem with DTMF or is it just certain carriers that have problems with DTMF? -mark On Jan 24, 2005, at 6:50 PM, Juan Cardenas wrote: I have experience that problem on numerous ocassions with Voicepulse Connect service using IAX for inbound service. DMTF times out or fails to read certain digits(tones). When had it configured to use SIP for incoming calls, it never failed. - Original Message - From: Mark Eissler [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com; Brian Dingman [EMAIL PROTECTED] Sent: Monday, January 24, 2005 3:06 PM Subject: Re: [Asterisk-Users] LiveVoip DTMF Issues Same problem I'm having with VP Connect. Perhaps it's a question of the version of Asterisk being run. I'm on 1.0.2. -mark On Jan 24, 2005, at 11:36 AM, Brian Dingman wrote: I have a couple of DID's with LiveVoip and am having major DTMF issues on incoming calls. I am connecting to them through IAX using ULAW. When someone dials one of these DD's (from a landline) they are for the most part unable to navigate the IVR menu successfuly. I would say the failure rate is greater than 80%. For example if the caller presses 5 sometimes * will see the DTMF as 55 or 555 or not see it at all. Is there anything I can do on my end to fix this problem, or is the old axim you get what you pay for true? It should also be noted that I have some other DID's from other providers and DTMF recognition is pretty much dead on. ___ 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 -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Terminiation in the UK.
Can somebody help me with cheap terminiation in the UK ? With different areacodes for in/out going traffic. Please contact me OFFLIST /Regards Mike ___ 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] LiveVoip DTMF Issues
On Tue, 25 Jan 2005 08:24:36 -0500, Mark Eissler wrote: My assumption is that most folks trunking through Voicepulse Connect must be using SIP since I haven't seen this problem mentioned before. So my conclusion is that DTMF and SIP and VPC work fine together BUT then you don't get to benefit from the efficiency of IAX. So the million dollar question is: Does IAX have a problem with DTMF or is it just certain carriers that have problems with DTMF? -mark On Jan 24, 2005, at 6:50 PM, Juan Cardenas wrote: I have experience that problem on numerous ocassions with Voicepulse Connect service using IAX for inbound service. DMTF times out or fails to read certain digits(tones). When had it configured to use SIP for incoming calls, it never failed. A while back when I used VPC I had not trouble with DTMF. I even used it over IAX for DISA, which requires DTMF for authentication. You just need to set the dtmfmode correctly, which varies by codec. Voipjet had a problem with dtmf right after they launched, but that was resolved quickly. Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 ___ 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 with Grandstream ringback
- Original Message - From: Kim Lux [EMAIL PROTECTED] To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, January 25, 2005 7:54 AM Subject: Re: [Asterisk-Users] Asterisk with Grandstream ringback I've got the same problem with the same firmware version. I also spot same behaviour - same under .16 and .18 v. Regards, Rob. On Mon, 2005-01-24 at 16:46 +0200, Doug Reid - Stormcorp wrote: Hi All We have Grandstream 102's running ver X.18. When hanging up after a call has been made the grandstream seems not to disconnect the call and when you put the handset down the phone rings only to pick it up and be on the same call. This is happening quite often and gets very irritating. Can anyone help with this? Regards Doug ___ 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 -- Kim Lux, Diesel Research Inc. ___ 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] Updating Asterisk
Just tried it. Show version still shows: Connected to Asterisk CVS-v1-0-12/21/04-14:14:46 Well, only thing I can see is that your CVS download didn't went right, or you downloaded it into a different place, because you're not even at 1.0.4 Follow these simple steps to update you tree : # cd /usr/src # export CVSROOT=:pserver:[EMAIL PROTECTED]:/usr/cvsroot # cvs login - the password is anoncvs. # cvs checkout -r v1-0-5 asterisk # cd asterisk # make clean; make then, stop asterisk # make install then start asterisk 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] Updating Asterisk
Adam Robins wrote: Just tried it. Show version still shows: Connected to Asterisk CVS-v1-0-12/21/04-14:14:46 Which is exactly what it said prior to the upgrade. in the asterisk src directory there is a .version file, remove this and it will update with the current time/date... -- Best regards, Duane http://www.cacert.org - Free Security Certificates http://www.nodedb.com - Think globally, network locally http://www.sydneywireless.com - Telecommunications Freedom http://happysnapper.com.au - Sell your photos over the net! http://e164.org - Using Enum.164 to interconnect asterisk servers I do not try to dance better than anyone else. I only try to dance better than 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] OT: pinout forstandard telephoneheadsetrequired.?
Many thanks Julian. Mike On Tue, 25 Jan 2005 11:30:59 +, Asterisk [EMAIL PROTECTED] wrote: I'm going into work now, and will send the specs when I'm there. Julian. Mike Dent wrote: Julian, can you remember how you wired it, ie which pin on the plantronics connector went to which pin on the ciso? Mike On Tue, 25 Jan 2005 10:32:50 +, Asterisk [EMAIL PROTECTED] wrote: Damn, that would have saved us a lot of time. We recently had to modify 55 headsets ... :( Julian. Florian Overkamp wrote: Hi, -Original Message- We found that the plantronics headet that we used to have for the Meridian phones did not work in the cisco headset jack. We had to cut the ends off and rewire them, which we worked out by trial and error. You can buy plantronics for cisco phones as well. Actually, we've worked with plantronics sets and they came with two wires, one that matches the handset plug of the cisco, and one that matches the headset plug of the cisco... YMMV I guess. Florian ___ 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 ___ 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] Updating Asterisk
http://lists.digium.com/pipermail/asterisk-users/2004-December/080514.html On Tue, 25 Jan 2005 08:58:46 -0500, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Just tried it. Show version still shows: Connected to Asterisk CVS-v1-0-12/21/04-14:14:46 Well, only thing I can see is that your CVS download didn't went right, or you downloaded it into a different place, because you're not even at 1.0.4 Follow these simple steps to update you tree : # cd /usr/src # export CVSROOT=:pserver:[EMAIL PROTECTED]:/usr/cvsroot # cvs login - the password is anoncvs. # cvs checkout -r v1-0-5 asterisk # cd asterisk # make clean; make then, stop asterisk # make install then start asterisk 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
RE: [Asterisk-Users] OT: pinout forstandardtelephoneheadsetrequired.?
Many thanks Julian. Are you looking for the pinout for a single plug 2.5mm (cellphone) headset or a dual plug 3.5mm (computer) headset? -- Nabeel Jafferali Tel: +1 (416) 628-9342 Toronto +1 (646) 225-7426 New York FWD: 46990 Email/MSN: nabeelatjafferali.net ___ 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] Updating Asterisk
There was discussion of this before... I thought: cvs checkout -r v1-0 would get you the latest stable version 1.0.X code On Tue, 25 Jan 2005 08:58:46 -0500, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Follow these simple steps to update you tree : # cd /usr/src # export CVSROOT=:pserver:[EMAIL PROTECTED]:/usr/cvsroot # cvs login - the password is anoncvs. # cvs checkout -r v1-0-5 asterisk # cd asterisk # make clean; make then, stop asterisk # make install then start asterisk 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] TE110P yellow errors
Hi All, I have a TE110P in E1 mode, in a dell poweredge 250. The 30 channel E1 supplied is from a telco in Australia, with the following in my zaptel.conf: span=1,1,0,ccs,hdb3 bchan=1-15 dchan=16 bchan=17-31 (yes the provider has NOcrc4 for some reason) After installing the latest cvs libpri and zaptel, I successfully loaded the kernel modules. The card is verified as not sharing an interrupt with anything else. The problem is I continuously get yellow errors, the IRQ missed counter goes up, and the light on the card blinks b/w red and green. I have verified the cabling, as it works fine in the E1 port of a cisco router below it. Any ideas most welcome thanks. TIA ___ 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] OT: pinout forstandardtelephoneheadsetrequired.?
Hi Neither, the one I am looking for is the tiny (similar to RJ11) plug. Which are used on telephony headsets. Mike On Tue, 25 Jan 2005 09:06:57 -0500, Nabeel Jafferali [EMAIL PROTECTED] wrote: Many thanks Julian. Are you looking for the pinout for a single plug 2.5mm (cellphone) headset or a dual plug 3.5mm (computer) headset? -- Nabeel Jafferali Tel: +1 (416) 628-9342 Toronto +1 (646) 225-7426 New York FWD: 46990 Email/MSN: nabeelatjafferali.net ___ 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] Codec mismatch between SIP (BT) and IAX Phone
(IAXPhone): I suppose you're talking about Steve Sokol's phone If so, then this phone only support gsm. Jan 25 13:52:22 NOTICE[31334]: channel.c:1314 ast_read: Dropping incompatible voice frame on IAX2/200/1 of format gsm since our native format has changed to ulaw Why is Asterisk not satisfied with gsm packets - it should transcode if necessary ? I have enabled gsm and ulaw in both configs, but it seems not sufficient. Yes, * will transcode, but you specified in the IAX Phone config that you allow this one tu use gsm AND ulaw, so instead of transcoding, * just tell the IAX Phone to switch to uLaw, since the originating party sends it in ulaw. Just change your iax.conf to only allow gsm on the IAX Phone like this : disallow=all allow=gsm ___ 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] Updating Asterisk
Duane wrote: Adam Robins wrote: Just tried it. Show version still shows: Connected to Asterisk CVS-v1-0-12/21/04-14:14:46 Which is exactly what it said prior to the upgrade. in the asterisk src directory there is a .version file, remove this and it will update with the current time/date... Heh, I was wondering if someone was going to point this out. I was about to post it myself! Pretty early morning for you Duane? :) -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ 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 HEAD - Stable schedule?
hi does anyone know when current HEAD is scheduled to stabilise? Is there a plan, or is it still some time in the future? roy ___ 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] LiveVoip DTMF Issues
Mark, I don't know what to tell you. With my DID's from VP Connect, DTMF works fine over IAX. Even one of the lines I have with LiveVoip seems OK over IAX. The other well... it really doesn't work at all. So what does this say about * and DTMF recognition over IAX? Or the service providers? On Tue, 25 Jan 2005 07:45:08 -0600, Michael Graves [EMAIL PROTECTED] wrote: On Tue, 25 Jan 2005 08:24:36 -0500, Mark Eissler wrote: My assumption is that most folks trunking through Voicepulse Connect must be using SIP since I haven't seen this problem mentioned before. So my conclusion is that DTMF and SIP and VPC work fine together BUT then you don't get to benefit from the efficiency of IAX. So the million dollar question is: Does IAX have a problem with DTMF or is it just certain carriers that have problems with DTMF? -mark ___ 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] OT: pinout forstandardtelephoneheadsetrequired.?
Mike Dent wrote: Neither, the one I am looking for is the tiny (similar to RJ11) plug. Which are used on telephony headsets. The RJ10. Well, http://www.mml.uni-hannover.de/einhorn/headset/index_e.html has the Cisco 7960 headset jack first. Then, later they have the handset jack, which I am pretty sure is the same as a standard telephone headset jack. You could try both - that's what I did when building my single plug 2.5mm (cellphone) headset to Cisco 7960 headset adaptor. -- Nabeel Jafferali Tel: +1 (416) 628-9342 Toronto +1 (646) 225-7426 New York FWD: 46990 Email/MSN: nabeelatjafferali.net ___ 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] PrivacyManager not Working
Keith, VP Connect is having issues right now with callerid being transmitted... as much as they don't want to believe it. Sometimes it works, sometimes it doesn't. Maybe this is part of the problem. Does PM not work 100% of the time for you? On Mon, 24 Jan 2005 21:29:37 -0500, Keith O'Brien [EMAIL PROTECTED] wrote: I have been having problems getting PrivacyManager to work correctly. Right now I am running the 1/21/05 CVS but I have been unable to get this to work on asterisk-stable either. You can see from the debug below that the inbound call is arriving via IAX2 and both the CALLING NUMBER and CALLING NAME are both set as Unavailable. However, PrivacyManager executes and determines that the CallerID is present: -- CallerID Present: Skipping Anyone have an idea as to why this isn't working? Bug? asterisk1*CLI Rx-Frame Retry[No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 00011ms SCall: 00335 DCall: 0 [66.234.228.170:4569] VERSION : 2 CALLED NUMBER : 7326556755 CALLING NUMBER : Unavailable ** CALLING NAME: Unavailable ** LANGUAGE: en USERNAME: voicepulse-in-01 FORMAT : 4 CAPABILITY : 1086 ADSICPE : 2 DATE TIME : 171511810 Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: AUTHREQ Timestamp: 00015ms SCall: 1 DCall: 00335 [66.234.228.170:4569] AUTHMETHODS : 4 CHALLENGE : 123344711 USERNAME: voicepulse-in-01 Rx-Frame Retry[No] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: AUTHREP Timestamp: 00049ms SCall: 00335 DCall: 1 [66.234.228.170:4569] RSA RESULT : Sc+mxi0AL1JdD4Gh3s8Y5LJ13MrLm4DNNMDkCV2a5nSwuPx9djbCr2YmJO7eoxCbrP+077fdeMhpfXo -- Accepting AUTHENTICATED call from 66.234.228.170, requested format = 4, actual format = 4 -- Executing PrivacyManager([EMAIL PROTECTED]:4569]/1, ) in new stack Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: ACCEPT Timestamp: 00051ms SCall: 1 DCall: 00335 [66.234.228.170:4569] FORMAT : 4 -- CallerID Present: Skipping ** -- Executing Dial([EMAIL PROTECTED]:4569]/1, SIP/5001) in new stack -- Called 5001 Extensions.conf === exten = 7326556755,1,PrivacyManager exten = 7326556755,2,DIAL(SIP/5001) exten = 7326556755,3,Voicemail(u5001) exten = 7326556755,4,Hangup ___ 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 auto-dial out with .call files: Can I provide caller ID to second extension ?
Hi, I'm setting up system with repeated calling of outside extension. When it answers, local extension will ring. Supplied caller id displays correctly on outside phone, but on local extension it's empty. Can I somehow supply proper caller id to local extension too ? Regards, 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] BUSY-tone on incoming calls?
On Tue, 25 Jan 2005, Tobias Jönsson wrote: No, PRI_CAUSE works great at least in 1.0.2, probably in the earlier 1.0 releases too. Busy() may play a busy tone to the caller instead of signalling busy so using PRI_CAUSE is much better in PRI or BRI environment. The behaviour of Busy() and Congestion() can be changed with the priindication setting in zapata.conf. The options are inband (default) or outofband. This only affects the two applications mentioned above. 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
Re: [Asterisk-Users] Asterisk - static nat - laptop w/siproxd - cisco 7960
On Jan 25, 2005, at 1:51 AM, Kim Lux wrote: I'm trying to get similar working with a Grandstream. I'm getting a lot of echo. My laptop is crashing when the call terminates. What are you using for the NAT setup on your laptop ? (firestarter) Are you adding any special rules to handle the SIP phone ? My laptop is running Mac OS X 10.3. By default when I share my internet connection, natd is doing the NAT but I have siproxd running so nat doesn't apply. Siproxd is responsible for rewriting to the packet. I'm wondering if we both have the same problem, ie outside entities can't get back through the NAT to the phone connected to the laptop. That's what I'm seeing. My SIP phone cannot register to my asterisk box through siproxd. I'm not sure if it's the phone or siproxd but it's not asterisk -- asterisk doesn't care. Thanks. On Mon, 2005-01-24 at 13:54 -0500, Jason Lixfeld wrote: Ok, I have a 7960 that's plugged into my laptop. my home network is wireless so I don't have a switch anywhere to plug the phone into directly. I'm running siproxd on my OS X laptop and I can make outbound calls from the 7960 fine (I guess I don't have the phone configured to register inbound calls via SIP), but the phone isn't registering to the asterisk box via siproxd and I can't figure out why.. [ Asterisk ] -- Net -- [ Router w/static nat for SIP RTP ] -- [ Laptop ] -- [ 7960 ] Anyone using a similar configuration? ___ 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 -- Kim Lux, Diesel Research Inc. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk - static nat - laptop w/siproxd - cisco 7960
On Jan 25, 2005, at 2:02 AM, Adam Goryachev wrote: On Mon, 2005-01-24 at 23:51 -0700, Kim Lux wrote: I'm trying to get similar working with a Grandstream. I'm getting a lot of echo. My laptop is crashing when the call terminates. What are you using for the NAT setup on your laptop ? (firestarter) Are you adding any special rules to handle the SIP phone ? I'm wondering if we both have the same problem, ie outside entities can't get back through the NAT to the phone connected to the laptop. Maybe I'm missing something, but why would you possibly want the laptop to do NAT? Isn't your router or something else already doing that for your laptop traffic? So, why not just get linux or mac osx or whatever to bridge the wifi + ethernet, and then your sip phone will just talk to the network like normal... Looked into that, but a lot of OS' have difficulties bridging ethernet and 802.11 interfaces. OS X is no different. Hence the reason for siproxd. Just my thoughts ... Regards, Adam ___ 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] Updating Asterisk
On Tue, 2005-01-25 at 08:08 -0600, [EMAIL PROTECTED] wrote: Just tried it. Show version still shows: Connected to Asterisk CVS-v1-0-12/21/04-14:14:46 Which is exactly what it said prior to the upgrade. I can help here-- I've done this. If you do a cvs update, you need to do a make clean before you do the make. My conclusion is that not all the dependencies are encoded in the Makefile, and therefore, it is far safer to make clean; make than not not to clean everything and just go ahead with the make. Some files may not be updated thereby, that really should be, and you will have unpredictable problems. This will also guarantee that your stored version is updated. murf signature.asc Description: This is a digitally signed message 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] Nufone and Dialing Out
On January 25, 2005 08:07 am, Eric Wieling wrote: If he's dialing via Dial(IAX2/username:[EMAIL PROTECTED] then I don't think Asterisk will use the peer entry anyway. That is exactly my point. He either needs to specify the NANPA context if doing it the way he is, or use [EMAIL PROTECTED]) and have a nufone peer entry which specifies the context. At least that is my current best guess as to why it's not working for him. :-) -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Directory() ringing problem
The Directory command is working properly but the ringing herd in the origination phone is either garbled or herd infrequently. The termination phone does ring with consistency. Any suggestion on what might be happening. Kurt ___ 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] Intermittent breakage with the ISDN4Linux modem driver
On Fri, 21 Jan 2005, Steve Hill wrote: Every so often the ISDN just stops working (it neither dials out nor accepts incoming calls). Trying to dial out just logs: -- Executing Dial(IAX2/[EMAIL PROTECTED]/1, Modem/g1/48:19) in new stack and then sits there. Ok, more on this (if anyone can help). At the end of an outgoing call the ISDN is staying in the channel list: Modem[i4l]/ttyI1 (internal s1 )Down (None) (None) IAX2/[EMAIL PROTECTED]/3 (main-pabx-dial h1 ) Up Dial Modem/g1/48:h Any help appreciated. - Steve Jabber: [EMAIL PROTECTED] Web: http://www.nexusuk.org/ Servatis a periculum, servatis a maleficum - Whisper, Evanescence ___ 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] Outbound analog dialing with Internet Line Jack (fwd)
I've been trying to setup asterisk with an Internet Line Jack card for sometime. I've been successful in configuring asterisk to handle incoming calls, make calls between sip phones, call the asterisk demo, and even When the call comes in, what's the channel * reports handling ? Something like Zap/1 maybe ? The channel that shows handling is Phone/Phone0. ___ 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 with Grandstream ringback
Hi All Has any one tested Ver X.22 on the grandstreams? If so have you noticed the problem experienced with ringback? When you hang up the GS rings again and its the same call you put down. Only seen this with Ver X.16 and X.18 not yet with X.22 but I'm still not 100% convinced. Doug ___ 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: [Fwd: Re: [Asterisk-biz] bellster.net
Hi, In France, the second most important ADSL provider (named Free) offers a phone line (which uses VoIP but can only be used as a FXS) with unlimited free calls to landlines. I was wondering if I would use my Free phone line with Bellster as well, but I am not sure this is authorized by the ISP : http://adsl.free.fr/hd/cgv.html [in French] En particulier, l'utilisation du service à d'autres fins que privative (par exemple partage de l'accès téléphonique avec des personnes extérieures au foyer) ou raisonnable (taux d'utilisation manifestement excessif pour un abonné particulier par exemple) ainsi que l'utilisation à titre gratuit ou onéreux du service téléphonique de Freebox en tant que passerelle de réacheminement de communications, est strictement prohibée. In english, it is an extract of the ToS saying : the sharing of the line with people outside of the family, or the usage of the line as a communication bridge/gateway is strictly prohibited. Once this is said, if the number of calls done by Bellster users is limited, what's the probability the ISP discovers the trick...? It's an open question, I don't know how the ISP would react... Guillaume ___ 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] Different EXT lines for different users?
Hello!
I would like to make asterisk to use different ISDN external lines
dependant on which internal user makes the call. Right now I have
(12345678 represents my MSN):
[pstn] ; ISDN to PSTN
exten = _0.,1,Dial(CAPI/12345678:b${EXTEN:1})
exten = _0.,2,Hangup
This ofcourse means that whenever someone call's out to number 0this
call goes to outside line 12345678.
Now i would like asterisk to behave like that:
When user 100 calls go outside on line 12345670
When user 101 calls go outside on line 12345671
...
How can I do that?
Thank you,
Alen
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Re: [Asterisk-Users] Cisco 7940/7960
Doug Lytle wrote: Mark Johnson wrote: This may be OT, but I can't seem to find how to do this. I have 7940/7960's with Skinny on them. When you start pressing numbers on the dialpad, you start building a number to dial. When I install SIP, that functionality goes away. You have to hit the speaker button, or lift the handset before you can start dialing. Is there a setting I am missing, or is this just a product of SIP and I have to live with? Mark, I just got a 7940(eBay) and put the 7.3 SIP image on it. To dial, I can either start dialing to build the number and press either the # key to initiate the dial or presss the dial option on the lcd panel. Doug I also have loaded POS3-07-3-00 and hitting any numbers does nothing. I am using the default dialplan.xml file and a really basic SIPxxx.cnf file. This is the same on a couple of phones I am trying. Any ideas? ___ 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] ISP connection to the PSTN using Asterisk
Sorry everybody...first post on the list and my mail client gives me a hard time. Didn't think that first one made 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] LiveVoip DTMF Issues
I have been using Voicepulse connect via IAX for quite a while and have not had a problem. An auto attendant answers so I believe I would know if there were issues with DTMF. On Tuesday 25 January 2005 06:24 am, Mark Eissler wrote: My assumption is that most folks trunking through Voicepulse Connect must be using SIP since I haven't seen this problem mentioned before. So my conclusion is that DTMF and SIP and VPC work fine together BUT then you don't get to benefit from the efficiency of IAX. So the million dollar question is: Does IAX have a problem with DTMF or is it just certain carriers that have problems with DTMF? -mark On Jan 24, 2005, at 6:50 PM, Juan Cardenas wrote: I have experience that problem on numerous ocassions with Voicepulse Connect service using IAX for inbound service. DMTF times out or fails to read certain digits(tones). When had it configured to use SIP for incoming calls, it never failed. - Original Message - From: Mark Eissler [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com; Brian Dingman [EMAIL PROTECTED] Sent: Monday, January 24, 2005 3:06 PM Subject: Re: [Asterisk-Users] LiveVoip DTMF Issues Same problem I'm having with VP Connect. Perhaps it's a question of the version of Asterisk being run. I'm on 1.0.2. -mark On Jan 24, 2005, at 11:36 AM, Brian Dingman wrote: I have a couple of DID's with LiveVoip and am having major DTMF issues on incoming calls. I am connecting to them through IAX using ULAW. When someone dials one of these DD's (from a landline) they are for the most part unable to navigate the IVR menu successfuly. I would say the failure rate is greater than 80%. For example if the caller presses 5 sometimes * will see the DTMF as 55 or 555 or not see it at all. Is there anything I can do on my end to fix this problem, or is the old axim you get what you pay for true? It should also be noted that I have some other DID's from other providers and DTMF recognition is pretty much dead on. ___ 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 -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.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 -- |- - - - - - - - - - - - - - - - - - - -| |-Mike Deweyof -| |= All Technologies Unlimited, Inc =| |- phone: 303.667.0357 -| |- e-mail: [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] Asterisk with Grandstream ringback
Are you saying that you are running firmware X.22 and it is not doing the callback when you hang up ? Where exactly did you get that firmware version ? Thanks On Tue, 2005-01-25 at 16:55 +0200, Doug Reid - Stormcorp wrote: Hi All Has any one tested Ver X.22 on the grandstreams? If so have you noticed the problem experienced with ringback? When you hang up the GS rings again and its the same call you put down. Only seen this with Ver X.16 and X.18 not yet with X.22 but I'm still not 100% convinced. Doug ___ 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 -- Kim Lux, Diesel Research Inc. ___ 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] iax.conf qualify=yes not working?
Brent Goran wrote: We have many IAXy devices in the field now. In all cases, in iax.conf, we have qualify=yes, so that using iax2 show peers, we can see whether or not the device is currently online. In some cases, the IAXy device and/or Asterisk are not communicating their qualification, because iax2 show peers shows the device as status UNKNOWN. However, when a user picks up the telephone plugged into the IAXy, they can place a call just fine within our Asterisk server. Can anyone tell me if there are any conditions which might affect the functioning of the qualify feature, while still allowing outbound calls to go through? Yes, if the iax endpoint doesn't respond to IAX POKE frames with a PONG. iaxclient-based softphones (and, actually, anything based on libiax2) didn't do this until about code went it about 2 weeks ago.. I guess the iaxy is loosely based on libiax2, and may not implement this. -SteveK ___ 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] Goto invalid extension doesn't go to 'I' when in a macro.
Hi,
A bit of a problem here which I'd appreciate some thoughts on.
(please excuse the stray capital letters - Outlook has a habit of
capitalising where I don't want it to!)
For various reasons, I need to be able to do the following:
--8--
[default]
Exten = s,1,Macro(dosomething,)
Exten = s,2,NoOp(Returned)
[macro-dosomething]
Exten = s,1,Goto(${ARG1},1)
Exten = _[123]XXX,1,NoOp(Success)
Exten = I,1,NoOp(Failure)
--8--
Which should allow me to trap an invalid entry from the caller.
If ${ARG1} contains, say, 2000, it all works fine. However, in the above
example where ${ARG1} contains , the macro just finishes and control is
returned to the calling context - i.e. the extension 'i' in
[macro-dosomething] never gets called.
This is completely different behaviour than when Goto() is used in a
non-macro context - e.g.:
--8--
[default]
Exten = s,1,Goto(,1)
Exten = _[123],1,NoOp(Success)
Exten = I,1,NoOp(Failure)
--8--
Where control is passed to 'I' as expected.
The only way I have found around this is to include another context to allow
me to have:
Exten = _[123],1,NoOp(Success)
And
Exten = _.,1,NoOp(Failure)
In the same context.
Is this a bug or intentional behaviour? Does anybody else have a fix for
this?
Cheers,
Nick.
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RE: [Asterisk-Users] bellster credits problem coming...
According to Ed Guy at Bellster The most specific routes takes precedence. For example, if you are calling 1-212-555-1212 first routes for 1-212-555 are checked, then 1-212, then 1 until a non-congested route is found. (The searching is actually a bit more general -- matching is done on a per digit basis to meet international needs, but I cant image why anyone would publish a route of 1-21) /ed PS. 1-XXX-555- is blocked. I just use that as an example. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Jay Austad Sent: Tuesday, January 25, 2005 1:26 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] bellster credits problem coming... I signed up for the FWD forums, but didn't receive my confirmation email. So, since the FWD guys read this, I though I'd post here. If you read the route report, +1 currently has 11720 available calls. If you look at the routes for specific area codes/prefixes, they all have a much smaller number of available calls each. How does Bellster determine what IAX trunk to try first? Does it round-robin all of the possible matches? Or, does it try to pick the most specific route and then gradually try less specific ones until one works? Given a round-robin or random type scenario, people like me who have very specific routes (612,651,952, and 763 area codes) are not going to get many calls routed through our systems, therefore we will have a very hard time accruing credits. People who offer routes to +1 are going to get an enormous number of credits and unintentionally hoard them by not possibly being able to use them all. People who offer routes to less used area codes can end up using all of their credits and being starved until a call randomly gets routed to them, even though they have in good faith offered up their system for use. Obviously trying more specific routes first is the better solution, but it still doesn't address the problem of people in infrequently called areas being starved for credits. For example, the 701 area code is ND. All calls between cities there are LD. So, my local calling area there in a small town might be 1701493. How many people will use Bellster to call a town of 600 people? There's no reward for someone in a small town to run it because even if someone did call the small town, the guys offering +1 routes are more likely to handle the call, and he'll never get any credits to use the system. Maybe there should be a credit donation feature, where you can donate a certain percentage or number of calls back into a pool that will get distributed evenly among people who handle few calls due to the neglect of the scheduling system or the fact that no one ever calls BFE, ND. Or maybe a weighting/precendence system would be better, where everyone on the network is assigned a precedence of say 1000. That number would get decremented for every minute (or a certain amount of time) they use the network, and also for time they are not even connected up to the network. When it reaches zero, they can't make calls. Time spent connected to the network will slowly regenerate their precedence, and calls they handle for others will more quickly regenerate. You could even use this to implement a queueing system, where if no lines are available because they are in use to a certain route, it puts them in a hold queue based on their precedence related to others in the queue waiting to put a call through, maybe even add a dialback feature so they don't have to wait on hold while the line is in use, when they pick up, they get some sort of message the line is available and press 1 to continue placing their call. Anyway, the basic point of this message is that there is currently not much incentive for people in remote/infrequently called areas to sign up. They will end up making their 10 calls and then be providing a service for others and not getting anything out of it. Additionally, it's dangerous to allow routes for toll-free numbers in the US. Some adult lines use toll-free numbers, but have a menu option to charge the call to your phone bill, even though it's not a 900 number. ~jay ___ 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] Updating Asterisk
And I just checked the ChangeLog file in /usr/src/asterisk and it show 1.0.5 -Original Message- From: Adam Robins Sent: Tuesday, January 25, 2005 10:13 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Updating Asterisk 1. I wiped out the /usr/src/asterisk directory structure 2. I followed the instructions below for re-downloading, installing and restarting Asterisk 3. The Asterisk module in /usr/sbin/asterisk reflects the new date/time Still shows version 1-0 12/21/2004. I can not find a .version file in the /usr/src/asterisk directory -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C F Sent: Tuesday, January 25, 2005 9:05 AM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Updating Asterisk http://lists.digium.com/pipermail/asterisk-users/2004-December/080514.ht ml On Tue, 25 Jan 2005 08:58:46 -0500, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Just tried it. Show version still shows: Connected to Asterisk CVS-v1-0-12/21/04-14:14:46 Well, only thing I can see is that your CVS download didn't went right, or you downloaded it into a different place, because you're not even at 1.0.4 Follow these simple steps to update you tree : # cd /usr/src # export CVSROOT=:pserver:[EMAIL PROTECTED]:/usr/cvsroot # cvs login - the password is anoncvs. # cvs checkout -r v1-0-5 asterisk # cd asterisk # make clean; make then, stop asterisk # make install then start asterisk 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 The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ 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] Codec negotiation
Hello On every Incoming SIP and IAX call I see the following in asterisk debug: Accepting AUTHENTICATED call from 10.10.10.10, requested format = gsm, requested prefs = (), actual format = g729, my prefs = (g729|gsm|g723|g726|ulaw|alaw) priority = mine The problem is that the codec preference on both parties is different The calling party has preference gsm/g729/etc The called party (the one you see this debug from) has preference g729/gsm/etc The problem is.. This call is now set up with G729... And I want it rather to be decided by the callING party (thus want the call to be negotiated GSM) What can I do about this? (I just want that if I receive a call the calling party decides the codec, and not my side) My IAX.conf and SIP.conf have the following allow settings now Allow=all Allow=g729 Allow=gsm Allow=ulaw Allow=alaw Help :-) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] .call file creation
Thanks much
Dan
On Tue, 25 Jan 2005, Glenn Powers wrote:
Dan Adams wrote:
I am curious partly because it has occurred randomly in my asterisk
system. How does one go about creating a .call file for placing a call
between two extensions/phones? I know this has been mentioned and is
probably in one of the wikis somewhere, but I am unsure exactally how to
go about doing it. Can anyone point me in the right direction.
http://voip-info.org/tiki-index.php?page=Asterisk%20auto-dial%20out
Here's a web interface (click-to-dial) for creating .call files (part of
XRMS, an open source CRM package):
?php
/*
*
* CTI / Asterisk Outdial XRMS Plugin v0.2
* uses asterisk from:
* http://www.asterisk.org/
*
* copyright 2004 Glenn Powers [EMAIL PROTECTED]
* Licensed Under the Open Software License v. 2.0
*
*/
/*
*
* If you are using the sip.conf based lookupCID,
* Be sure to add crm_username(s) to your sip.conf file. See below.
*
* IF Asterisk is running on the same server as XRMS,
* MAKE SURE /var/spool/asterisk/outgoing is writable
* by your web server.
*
* IF asterisk is running on another server, use sftp
* to copy the file over.
*
*/
/*
* LookupCID :: ismaeljcarlo
* simple function to lookup extension number from sip.conf
*
* [EMAIL PROTECTED] created this function which looks up
* the value of crm_username from sip.conf and returns the proper extension.
*
*/
function lookupCID($thelookupCID) {
$lookupCID_sip_array = parse_ini_file(/etc/asterisk/sip.conf, true);
while ($v = current($lookupCID_sip_array)) {
if (isset($v['crm_username'])){
if($v['crm_username'] == $thelookupCID) {
$thelookupCID = key($lookupCID_sip_array);
return $thelookupCID;
}
}
next($lookupCID_sip_array);
}
}
/*
* End LookupCID
*/
// include the common files
require_once('../../include-locations.inc');
require_once($include_directory . 'vars.php');
require_once($include_directory . 'utils-interface.php');
require_once($include_directory . 'utils-misc.php');
require_once($include_directory . 'adodb/adodb.inc.php');
require_once($include_directory . 'adodb-params.php');
$con = adonewconnection($xrms_db_dbtype);
$con-connect($xrms_db_server, $xrms_db_username, $xrms_db_password,
$xrms_db_dbname);
// $con-debug = 1;
$session_user_id = session_check();
$session_username = $_SESSION['username'];
$msg = $_GET['msg'];
$contact_id = $_GET['contact_id'];
$company_id = $_GET['company_id'];
$phone = $_GET['phone'];
$phone_dial_prefix = 1;
$msg = urlencode(_(Dialing Phone Number: ) . $phone);
// Get contact name
$sql = SELECT first_names,last_name from contacts
WHERE contact_id = . $contact_id . LIMIT 1;
$rst = $con-execute($sql);
if ($rst) {
if (!$rst-EOF) {
$contact_name = urlencode($rst-fields['first_names'] .
. $rst-fields['last_name']);
}
}
// Get variables from the custom fields of the user's contact id.
$sql = SELECT custom1, custom2, custom3 from contacts, users
WHERE users.user_id = . $session_user_id .
AND contacts.contact_id = users.user_contact_id
LIMIT 1;
$rst = $con-execute($sql);
if ($rst) {
if (!$rst-EOF) {
$channel = $rst-fields['custom1'];
$extension_to_dial = $rst-fields['custom2'];
$CID = $rst-fields['custom3'];
}
}
// $sipCID = lookupCID($session_username);
// This is the file that will be passed to Asterisk
$dial_file_contents = Channel:$channel$extension_to_dial
MaxRetries: 1
RetryTime: 60
WaitTime: 30
Callerid: $CID
Context: xrms
Extension: $phone_dial_prefix$phone
Priority: 1
;
$filename = $xrms_file_root . /tmp/outdial-$phone;
if (!$handle = fopen($filename, 'w')) {
echo Cannot open file ($filename);
exit;
}
if (fwrite($handle, $dial_file_contents) === FALSE) {
echo Cannot write to file ($filename);
exit;
}
system(mv $filename /var/spool/asterisk/outgoing);
fclose($handle);
// Create an Activity on Dial
header(Location: ../../activities/new-2.php?user_id= . $session_user_id
. activity_status=oactivity_type_id=1contact_id=
. $contact_id . company_id= . $company_id . activity_title=
. _(Call%20To%20) . $contact_name
. return_url=/contacts/one.php?contact_id= . $contact_id);
// if you don't want to create an activity on dial, use this instead:
// header(Location:
$http_site_root/contacts/one.php?contact_id=$contact_idmsg
if (fwrite($handle, $dial_file_contents) === FALSE) {
echo Cannot write to file ($filename);
exit;
}
system(mv $filename /var/spool/asterisk/outgoing);
fclose($handle);
// Create an Activity on Dial
header(Location: ../../activities/new-2.php?user_id= . $session_user_id
. activity_status=oactivity_type_id=1contact_id=
. $contact_id . company_id= . $company_id . activity_title=
. _(Call%20To%20) . $contact_name
. return_url=/contacts/one.php?contact_id= . $contact_id);
// if you don't want to create an activity on dial,
RE: [Asterisk-Users] bellster credits problem coming...
Jay, Thanks for the feedback. You seem to be missing one of the basic premises of bellster: it is an equitable sharing network where The [Calls] you take are equal to the [Calls] you make. Route selection is done heuristically favoring the least used of the most direct (or more fully specified) routes. Many routes are attempted until the call is successfully routed. It is neither round-robin or random. For instance, if Marge Gunderson in Fargo runs the only bellster node for her small exchange in North Dakota, calls to that exchange will go there first, then if there is no PSTN path available, it attempt higher level routes (e.g., the area code, then the country) until a working one is found. I'll add these features as schedule permits: * Altruistic Routes where the caller need not have any credits to call. * Points Transfer On the chargeable 800 numbers, please provide specific details off-list. /ed guy [EMAIL PROTECTED] PS. recent features: * Quiet Calls. (sans Allison) * ENUM directory. (server side is done -- hopefully someone will donate the client side.) see http://www.bellster.net/web/NewFeatures PPS. for your FWD mailing list problem, visit support at: http://www.fwdnet.net/content/view/full/373/ -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Jay Austad Sent: Tuesday, January 25, 2005 1:26 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] bellster credits problem coming... I signed up for the FWD forums, but didn't receive my confirmation email. So, since the FWD guys read this, I though I'd post here. If you read the route report, +1 currently has 11720 available calls. If you look at the routes for specific area codes/prefixes, they all have a much smaller number of available calls each. How does Bellster determine what IAX trunk to try first? Does it round-robin all of the possible matches? Or, does it try to pick the most specific route and then gradually try less specific ones until one works? Given a round-robin or random type scenario, people like me who have very specific routes (612,651,952, and 763 area codes) are not going to get many calls routed through our systems, therefore we will have a very hard time accruing credits. People who offer routes to +1 are going to get an enormous number of credits and unintentionally hoard them by not possibly being able to use them all. People who offer routes to less used area codes can end up using all of their credits and being starved until a call randomly gets routed to them, even though they have in good faith offered up their system for use. Obviously trying more specific routes first is the better solution, but it still doesn't address the problem of people in infrequently called areas being starved for credits. For example, the 701 area code is ND. All calls between cities there are LD. So, my local calling area there in a small town might be 1701493. How many people will use Bellster to call a town of 600 people? There's no reward for someone in a small town to run it because even if someone did call the small town, the guys offering +1 routes are more likely to handle the call, and he'll never get any credits to use the system. Maybe there should be a credit donation feature, where you can donate a certain percentage or number of calls back into a pool that will get distributed evenly among people who handle few calls due to the neglect of the scheduling system or the fact that no one ever calls BFE, ND. Or maybe a weighting/precendence system would be better, where everyone on the network is assigned a precedence of say 1000. That number would get decremented for every minute (or a certain amount of time) they use the network, and also for time they are not even connected up to the network. When it reaches zero, they can't make calls. Time spent connected to the network will slowly regenerate their precedence, and calls they handle for others will more quickly regenerate. You could even use this to implement a queueing system, where if no lines are available because they are in use to a certain route, it puts them in a hold queue based on their precedence related to others in the queue waiting to put a call through, maybe even add a dialback feature so they don't have to wait on hold while the line is in use, when they pick up, they get some sort of message the line is available and press 1 to continue placing their call. Anyway, the basic point of this message is that there is currently not much incentive for people in remote/infrequently called areas to sign up. They will end up making their 10 calls and then be providing a service for others and not getting anything out of it. Additionally, it's dangerous to allow routes for toll-free numbers in the US. Some adult lines use toll-free numbers, but have a menu option to charge the call to your phone bill, even though it's not a 900 number. ~jay
RE: [Asterisk-Users] Cisco 7940/7960
We use the 7690 and it works fine there. Has nothing to do with SIP as Snom, ACT, 7960 ect all work that way. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Mark Johnson Sent: Tuesday, January 25, 2005 5:04 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Cisco 7940/7960 Doug Lytle wrote: Mark Johnson wrote: This may be OT, but I can't seem to find how to do this. I have 7940/7960's with Skinny on them. When you start pressing numbers on the dialpad, you start building a number to dial. When I install SIP, that functionality goes away. You have to hit the speaker button, or lift the handset before you can start dialing. Is there a setting I am missing, or is this just a product of SIP and I have to live with? Mark, I just got a 7940(eBay) and put the 7.3 SIP image on it. To dial, I can either start dialing to build the number and press either the # key to initiate the dial or presss the dial option on the lcd panel. Doug I also have loaded POS3-07-3-00 and hitting any numbers does nothing. I am using the default dialplan.xml file and a really basic SIPxxx.cnf file. This is the same on a couple of phones I am trying. Any ideas? ___ 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] BackupPc_nightly crashing with Perl chdir errors
Hi all. I have been reading the list archives and can't seem to find
anything that relates to these errors.
BackupPC_nightly fails to delete any files, and reports the pool and cpool
at zero size. The nightly run as currently configured should be deleting
a ton of files (only one hardlink) but it deletes nothing, and so the
drive is now 100% and staying there. I have adjusted the conf files on
the backupjobs so a majority of the pool would be deleted in a proper
nightly run, but as listed below nothing is removed.
Reports in the log are as follows:
2005-01-25 01:30:00 Running 4 BackupPC_nightly jobs from 0..15 (out of 0..15)
2005-01-25 01:30:01 Running BackupPC_nightly -m 0 63 (pid=369)
2005-01-25 01:30:01 Running BackupPC_nightly 64 127 (pid=370)
2005-01-25 01:30:01 Running BackupPC_nightly 128 191 (pid=371)
2005-01-25 01:30:01 Running BackupPC_nightly 192 255 (pid=372)
2005-01-25 01:30:01 Next wakeup is 2005-01-25 02:00:00
2005-01-25 01:30:01 admin : Use of uninitialized value in chdir at
/usr/lib/perl5/5.8.4/File/Find.pm line 741.
2005-01-25 01:30:01 admin : Use of chdir('') or chdir(undef) as chdir()
is deprecated at /usr/lib/perl5/5.8.4/File/Find.pm line 741.
2005-01-25 01:30:01 admin : Use of uninitialized value in concatenation
(.) or string at /usr/lib/perl5/5.8.4/File/Find.pm line 742.
2005-01-25 01:30:01 admin : Can't cd to : Permission denied
2005-01-25 01:30:01 Finished admin (BackupPC_nightly -m 0 63)
2005-01-25 01:30:19 admin2 : Use of uninitialized value in chdir at
/usr/lib/perl5/5.8.4/File/Find.pm line 741.
2005-01-25 01:30:19 admin2 : Use of chdir('') or chdir(undef) as chdir()
is deprecated at /usr/lib/perl5/5.8.4/File/Find.pm line 741.
2005-01-25 01:30:19 admin2 : Use of uninitialized value in concatenation
(.) or string at /usr/lib/perl5/5.8.4/File/Find.pm line 742.
2005-01-25 01:30:19 admin2 : Can't cd to : Permission denied
2005-01-25 01:30:19 Finished admin2 (BackupPC_nightly 128 191)
2005-01-25 01:30:30 admin1 : Use of uninitialized value in chdir at
/usr/lib/perl5/5.8.4/File/Find.pm line 741.
2005-01-25 01:30:30 admin1 : Use of chdir('') or chdir(undef) as chdir()
is deprecated at /usr/lib/perl5/5.8.4/File/Find.pm line 741.
2005-01-25 01:30:30 admin1 : Use of uninitialized value in concatenation
(.) or string at /usr/lib/perl5/5.8.4/File/Find.pm line 742.
2005-01-25 01:30:30 admin1 : Can't cd to : Permission denied
2005-01-25 01:30:30 Finished admin1 (BackupPC_nightly 64 127)
2005-01-25 01:30:32 admin3 : Use of uninitialized value in chdir at
/usr/lib/perl5/5.8.4/File/Find.pm line 741.
2005-01-25 01:30:32 admin3 : Use of chdir('') or chdir(undef) as chdir()
is deprecated at /usr/lib/perl5/5.8.4/File/Find.pm line 741.
2005-01-25 01:30:32 admin3 : Use of uninitialized value in concatenation
(.) or string at /usr/lib/perl5/5.8.4/File/Find.pm line 742.
2005-01-25 01:30:32 admin3 : Can't cd to : Permission denied
2005-01-25 01:30:32 Finished admin3 (BackupPC_nightly 192 255)
2005-01-25 01:30:32 Pool nightly clean removed 0 files of size 0.00GB
2005-01-25 01:30:32 Pool is 0.00GB, 0 files (0 repeated, 0 max chain, 0
max links), 1 directories
2005-01-25 01:30:32 Cpool nightly clean removed 0 files of size 0.00GB
2005-01-25 01:30:32 Cpool is 0.00GB, 0 files (0 repeated, 0 max chain, 0
max links), 0 directories
When I run a BackupPC_nightly when su'd as the backuppc user similar
results as follows:
backuppc root $ /usr/local/backuppc/bin/BackupPC_nightly 0 255
Use of uninitialized value in chdir at /usr/lib/perl5/5.8.5/File/Find.pm
line 741.
Use of chdir('') or chdir(undef) as chdir() is deprecated at
/usr/lib/perl5/5.8.5/File/Find.pm line 741.
Use of uninitialized value in concatenation (.) or string at
/usr/lib/perl5/5.8.5/File/Find.pm line 742.
Can't cd to : No such file or directory
I have been using backuppc for months with reliable service, and we love
the package. One of my Jr. Admins got a little trigger happy with gentoo
portage and upgraded everything on a backuppc server that had been working
pretty well. We also had an unrelated issue that caused the backup size
to dramatically increase and fill the drives, so I am not sure which is to
blame here.
I have partimaged images of the partition that holds the pool data, and a
few versions of the rsyncd root of the partition with the linux install on
it, but I would rather figure out how to solve this problem than roll back
time on the server and lose the backup data we have on the drives now.
I cannot seem to find anything in the docs or the lists to point to any
solution, any input would be greatly appreciated.
--
Thanks,
Michael McKinsey
FlashByte Digital Publishing
http://www.flashbyte.us
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Re: [Asterisk-Users] OT: pinout forstandardtelephoneheadsetrequired.?
Nabeel, I would be interested in seeing your schematic for the headset to 2.5 mm adaptor. On a side note ... anyone know if the Polycom headset jack uses the same pinouts??? thanks mike On Tuesday 25 January 2005 07:16 am, Nabeel Jafferali wrote: Mike Dent wrote: Neither, the one I am looking for is the tiny (similar to RJ11) plug. Which are used on telephony headsets. The RJ10. Well, http://www.mml.uni-hannover.de/einhorn/headset/index_e.html has the Cisco 7960 headset jack first. Then, later they have the handset jack, which I am pretty sure is the same as a standard telephone headset jack. You could try both - that's what I did when building my single plug 2.5mm (cellphone) headset to Cisco 7960 headset adaptor. -- Nabeel Jafferali Tel: +1 (416) 628-9342 Toronto +1 (646) 225-7426 New York FWD: 46990 Email/MSN: nabeelatjafferali.net ___ 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 -- |- - - - - - - - - - - - - - - - - - - -| |-Mike Deweyof -| |= All Technologies Unlimited, Inc =| |- phone: 303.667.0357 -| |- e-mail: [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] Updating Asterisk
Adam Robins wrote: 1. I wiped out the /usr/src/asterisk directory structure 2. I followed the instructions below for re-downloading, installing and restarting Asterisk 3. The Asterisk module in /usr/sbin/asterisk reflects the new date/time Still shows version 1-0 12/21/2004. I can not find a .version file in the /usr/src/asterisk directory -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C F Sent: Tuesday, January 25, 2005 9:05 AM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Updating Asterisk http://lists.digium.com/pipermail/asterisk-users/2004-December/080514.ht ml On Tue, 25 Jan 2005 08:58:46 -0500, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Just tried it. Show version still shows: Connected to Asterisk CVS-v1-0-12/21/04-14:14:46 Well, only thing I can see is that your CVS download didn't went right, or you downloaded it into a different place, because you're not even at 1.0.4 Follow these simple steps to update you tree : # cd /usr/src # export CVSROOT=:pserver:[EMAIL PROTECTED]:/usr/cvsroot # cvs login - the password is anoncvs. # cvs checkout -r v1-0-5 asterisk # cd asterisk # make clean; make then, stop asterisk # make install then start asterisk Silly question.. Are you restarting asterisk? Are you sure? A reload won't do it. A restart now (which will distrupt traffic) should do it, but ultimately a stop now and start it back up would be better. By the way, how are you starting asterisk? If you are using a script, check what binary it points to. -Brett ___ 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] AMP with SUSE 9.2
Title: AMP with SUSE 9.2 Hi, I have the newbie guide from AMPs website and (fair enough) it is all about whitebox linux. Has anyone found any gotchas with the newbie guide relating to SUSE 9.2 ? Any help appreciated. Cheers ___ 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 7940/7960
On Tue, 25 Jan 2005 09:08:51 -0500, Mark Johnson [EMAIL PROTECTED] wrote: This may be OT, but I can't seem to find how to do this. I have 7940/7960's with Skinny on them. When you start pressing numbers on the dialpad, you start building a number to dial. When I install SIP, that functionality goes away. You have to hit the speaker button, or lift the handset before you can start dialing. Is there a setting I am missing, or is this just a product of SIP and I have to live with? Thanks! Craig Unfortunately this hot keypad functionality is not included with the 7940/7960 SIP image. It is on the 7905/7912 SIP image though. As I have both types of phones in use here, it's somewhat annoying having to adjust my dialing habits depending on the phone I'm using :-) *shrugs* -Shaun ___ 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 7940/7960
I also have loaded POS3-07-3-00 and hitting any numbers does nothing. I am using the default dialplan.xml file and a really basic SIPxxx.cnf file. This is the same on a couple of phones I am trying. Any ideas? I am running SIP image 6.3 on Cisco 7940. The same here, I have to pickup the receiver or hit the speaker before I can dial any numbers. On a 7940 with SCCP image I can simply hit start dialing a number, on a 7940 with SIP I cannot. Adi ___ 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 7940/7960
Doug Reid - Stormcorp wrote: We use the 7690 and it works fine there. Has nothing to do with SIP as Snom, ACT, 7960 ect all work that way. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Mark Johnson Sent: Tuesday, January 25, 2005 5:04 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Cisco 7940/7960 Doug Lytle wrote: Mark Johnson wrote: This may be OT, but I can't seem to find how to do this. I have 7940/7960's with Skinny on them. When you start pressing numbers on the dialpad, you start building a number to dial. When I install SIP, that functionality goes away. You have to hit the speaker button, or lift the handset before you can start dialing. Is there a setting I am missing, or is this just a product of SIP and I have to live with? Mark, I just got a 7940(eBay) and put the 7.3 SIP image on it. To dial, I can either start dialing to build the number and press either the # key to initiate the dial or presss the dial option on the lcd panel. Doug I also have loaded POS3-07-3-00 and hitting any numbers does nothing. I am using the default dialplan.xml file and a really basic SIPxxx.cnf file. This is the same on a couple of phones I am trying. Any ideas? To the Dougs, This is turning into a me too, but my phones, about 25 of them, don't let me dial without picking up the handset, pressing the speaker button or the headset button or a line button. I cannot dial from the idle screen. I'm running the 7.3 SIP image, an absolutely barren dialplan.xml and basic SIP.cnf. Exactly what did you do to make the phone let you dial from the idle screen. -Chris ___ 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] iax java client
Hello. I am looking for a iax java client which could be used with our interface written in java to make iax connections with asterisk. Does anyone know something we could use? Thank you. Regards. ___ 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 HEAD - Stable schedule?
Roy Sigurd Karlsbakk wrote: does anyone know when current HEAD is scheduled to stabilise? Is there a plan, or is it still some time in the future? I believe I saw an announcement recently that it will start stabilizing in February, with the goal of releasing 1.1 on the six-month anniversary of the 1.0 release. ___ 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] SIP-T Support (I got my head in an SS7 cloud)
[EMAIL PROTECTED] wrote: But I would like asterisk to accept the IMTs.. and the only way to do that is to find out what DS0 to take and place the calls on based on the SS7 messages. I guess one of the things I'm not clear on is going from SS7 to SIP-T, I'm not sure where the state machine exists.. If that's the case, then you external box is just an SS7-SIP-T translator, right? It's not involved in the media path at all. It seems to me that if Asterisk is going to be involved in setup/teardown of the DS0s, then it needs to be involved in the signaling as well. Probably the best way to achieve this is going to be for Asterisk to support SIGTRAN (or one of the other SS7-over-IP solutions) and use an external SS7-to-SIGTRAN translator (or get SIGTRAN from your telco(s), if they support 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] AMP with SUSE 9.2
Keith Burns wrote: *Hi,* *I have the newbie guide from AMPs website and (fair enough) it is all about whitebox linux.** Has anyone found any gotchas with the newbie guide relating to SUSE 9.2 ?* Please post to the amportal mailing list: http://lists.sourceforge.net/lists/listinfo/amportal-users or Help forum: http://sourceforge.net/forum/?group_id=121515 SUSE does some things differently - the main difference is the apache2 (httpd) configuration. Regards, -- Jason Becker Director CEO Coalescent Systems Inc. 403.244.8089 www.coalescentsystems.ca ___ 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] TDM400 in aging Dell Optiplex
Ronan Mullally wrote: I've got an old Dell Optiplex (Pentium-II, 1998 Vintage) which is successfully running an X100P card. I'm hoping to upgrade to a TDM400. Has anybody tried running these cards in old Optiplex machines? I'm not particularly worried about horsepower - more about the motherboard having a PCI bus that's able to power up the card... -Ronan ___ 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 Got one running in an Optiplex GX100. Works fine. Jeff ___ 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] Dialing Delay
Like manxpower said, set DigitTimeout to 2 seconds or whatever u want. Visit voip-info and look for urself. All whats happening is that it waiting to see if u will press another number (pattern matching) by default digitstimeout is set to 6 seconds you might want to change your dialplan as well. Example: exten = 123,1,Dial(Zap/1| 555) Since extension numbers have to be unique u can set an ext. # 123 to dial the 555 number. ___ 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] Your Acerbic Tyrant will be off line for about 10 days
Greetings List, I know many of you are looking for advice from me but I am moving from the 28th until about the 4th of February. As moving does not always go as planned so I am letting you know that I may be out of internet touch for 10 days during the move depending on the closing and the Cable Modem guy. In case any cares to know, I am moving from South Florida to Asheville. I will try to check mail often but please do not think I am being rude if I do not answer for a while. Race Vanderdecken ___ 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] x-lite with wireless connection
Hello This might not be a 'pure' * question, but it is relevant to general VOIP technology. I tried x-lite on my notebook with wireless connection(802.11). The software has been tested with the fixed line connection. It worked fine to call through *. When using wireless connection, it is clear on my side using notebook; however, there is loud noise on the other side of the call which uses IPphone. It seems to me that some interference noise comes into the upstream. Does anyone notice the same problem? or have explanation of the cause? regards, steven ___ 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] SIP-T Support (I got my head in an SS7 cloud)
I, personally, think a channel driver handling both sigtran and mgcp/megaco would be an ideal setup for bridging the gap between ip and pstn.. especially with the current hardware devices on the market.. but all of that is just opinion.. Kevin P. Fleming wrote: [EMAIL PROTECTED] wrote: But I would like asterisk to accept the IMTs.. and the only way to do that is to find out what DS0 to take and place the calls on based on the SS7 messages. I guess one of the things I'm not clear on is going from SS7 to SIP-T, I'm not sure where the state machine exists.. If that's the case, then you external box is just an SS7-SIP-T translator, right? It's not involved in the media path at all. It seems to me that if Asterisk is going to be involved in setup/teardown of the DS0s, then it needs to be involved in the signaling as well. Probably the best way to achieve this is going to be for Asterisk to support SIGTRAN (or one of the other SS7-over-IP solutions) and use an external SS7-to-SIGTRAN translator (or get SIGTRAN from your telco(s), if they support 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 begin:vcard fn:Matt Hess n:Hess;Matt org:LiveWireNet adr;dom:;;4577 Pecos St;Denver;CO;80211 email;internet:[EMAIL PROTECTED] title:Senior Network Engineer tel;work:303-458-5667 tel;fax:303-458-5725 x-mozilla-html:FALSE url:http://www.livewirenet.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] AMP with SUSE 9.2
Cool, will do, thanks! -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Jason Becker Sent: Tuesday, January 25, 2005 9:24 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] AMP with SUSE 9.2 Keith Burns wrote: *Hi,* *I have the newbie guide from AMP**'**s website and (fair enough) it is all about whitebox linux.** Has anyone found any gotchas with the newbie guide relating to SUSE 9.2 ?* Please post to the amportal mailing list: http://lists.sourceforge.net/lists/listinfo/amportal-users or Help forum: http://sourceforge.net/forum/?group_id=121515 SUSE does some things differently - the main difference is the apache2 (httpd) configuration. Regards, -- Jason Becker Director CEO Coalescent Systems Inc. 403.244.8089 www.coalescentsystems.ca ___ 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] Updating Asterisk
I initially installed Asterisk from a vendor supplied CD. I want to maintain a more current release so I am trying to update from CVS. I removed the previous vendor's release and followed the instructions you provided. I got the following error. Can you explain why? Thanks, Steve --- cut here --- isk/asterisk.conf\ -DASTMODDIR=\/usr/lib/asterisk/modules\ -DASTAGIDIR=\/var/lib/asterisk/agi-bin\ -DBUSYDETECT_MARTIN -c -o channel.o channel.c channel.c:49:2: #error You need newer zaptel! Please cvs update zaptel channel.c: In function `ast_channel_alloc': channel.c:303: `ZT_TIMERPONG' undeclared (first use in this function) channel.c:303: (Each undeclared identifier is reported only once channel.c:303: for each function it appears in.) channel.c: In function `ast_queue_frame': channel.c:418: `ZT_TIMERPING' undeclared (first use in this function) channel.c: In function `ast_read': channel.c:1244: `ZT_EVENT_TIMER_EXPIRED' undeclared (first use in this function) channel.c:1246: `ZT_EVENT_TIMER_PING' undeclared (first use in this function) channel.c:1255: `ZT_TIMERPONG' undeclared (first use in this function) make: *** [channel.o] Error 1 --- end cut -- [EMAIL PROTECTED] wrote: Just tried it. Show version still shows: Connected to Asterisk CVS-v1-0-12/21/04-14:14:46 Well, only thing I can see is that your CVS download didn't went right, or you downloaded it into a different place, because you're not even at 1.0.4 Follow these simple steps to update you tree : # cd /usr/src # export CVSROOT=:pserver:[EMAIL PROTECTED]:/usr/cvsroot # cvs login - the password is anoncvs. # cvs checkout -r v1-0-5 asterisk # cd asterisk # make clean; make then, stop asterisk # make install then start asterisk 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 -- ISC Network Engineering The University of Pennsylvania 3401 Walnut Street, Suite 221A Philadelphia, PA 19104 voice: 215-573-8396 215-746-8001 fax: 215-898-9348 sip:[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] Asterisk HEAD - Stable schedule?
[EMAIL PROTECTED] wrote: Roy Sigurd Karlsbakk wrote: does anyone know when current HEAD is scheduled to stabilise? Is there a plan, or is it still some time in the future? I believe I saw an announcement recently that it will start stabilizing in February, with the goal of releasing 1.1 on the six-month anniversary of the 1.0 release. I seem to recall somewhere that the thinking is that we'll be going with Linux-type nomenclature, where the even-numbered releases are STABLE, and the odd are HEAD. So 1.0.x STABLE will become 1.2.0 STABLE, and 1.1.x HEAD will be continued as 1.3.0 HEAD Could be wrong, but it'd make sense. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.7.2 - Release Date: 21/01/2005 ___ 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] coredumping on MusicOnHold
Hello, I have upgraded to 1.0.4 version of asterisk. After that asterisk crash every time On receiving an call from iax2 trunk to musiconhold application. SIP calls to MusicOnHold is however working. I already upgraded to 1.0.5, but the problem still Remainig. Any idea ? Iax2 : call proceding : Jan 25 17:29:40 DEBUG[9997]: pbx.c:1261 pbx_extension_helper: Launching 'WaitMusicOnHold' -- Executing WaitMusicOnHold(IAX2/[EMAIL PROTECTED]/3, 201) in new stack Jan 25 17:29:40 DEBUG[9997]: channel.c:1551 ast_prod: Prodding channel 'IAX2/[EMAIL PROTECTED]/3' Urgent handler Ouch ... error while writing audio data: : Broken pipe Sip : call proceding : Jan 25 17:34:04 DEBUG[10020]: pbx.c:1261 pbx_extension_helper: Launching 'WaitMusicOnHold' -- Executing WaitMusicOnHold(SIP/192.168.1.38-082257a0, 201) in new stack Jan 25 17:34:04 DEBUG[10020]: channel.c:1551 ast_prod: Prodding channel 'SIP/192.168.1.38-082257a0' Jan 25 17:34:04 DEBUG[10020]: channel.c:1707 ast_set_write_format: Set channel SIP/192.168.1.38-082257a0 to write format slin -- Started music on hold, class 'default', on SIP/192.168.1.38-082257a0 Jan 25 17:34:04 DEBUG[10020]: channel.c:1128 ast_settimeout: Scheduling timer at 160 sample intervals Urgent handler Jan 25 17:34:04 DEBUG[10020]: channel.c:1379 ast_read: Generator got voice, switching to phase locked mode Jan 25 17:34:04 DEBUG[10020]: channel.c:1128 ast_settimeout: Scheduling timer at 0 sample intervals Jan 25 17:34:04 DEBUG[10020]: rtp.c:1188 ast_rtp_write: Ooh, format changed from unknown to alaw Radovan ___ 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
