[asterisk-users] No application 'ReceiveFAX'
Hi! Have probably not understand how fax is working in Asterisk 1.6. I did install: ptlib-v1_12_0 h323plus-v1_19_7 dahdi-linux-complete-2.2.0.2+2.2.0 spandsp-0.0.5 asterisk-1.6.2 asterisk-addons-1.6.2 make menuselect in asterisk-1.6.2 source directory shows: [*] app_fax But core show applications doesnt show me any fax applications and when I try to receive a fax: exten = 960,1,Answer() exten = 960,2,Wait(3) exten = 960,3,ReceiveFAX(/var/spool/asterisk/tmp/fax.tiff) [Nov 30 09:29:18] WARNING[19893]: pbx.c:3677 pbx_extension_helper: No application 'ReceiveFAX' for extension (inputinterior.se, 960, 3) Can any guru guide me what I am doing wrong? Best regards, MAGNUS BENNGRD Direktnr 031-799 89 75 Fältspatsgatan 2 421 30 Västra Frölunda Tel. 031-799 89 00 Fax 031-799 89 01 www.inputinterior.se [1] Links: -- [1] http://www.inputinterior.se inline: 7___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Prevent Dial if any extension is busy
Leif Neland wrote:
But my problem comes when I speak on 0317998985 and someone calls on
985, the call
get to my celluar phone and ofc the other way around.
Is there a way to check if any extension is busy and in that case
jump to VoiceMail(0317998...@inputinterior.se,b)?
If both phones were directly connected sip, it could be done.
The problem is that you can't determine if the cellular is busy before
you call it.
...
The other option is to modify the source, and add an option to the
dial-command, to exit if any extension dialled is busy.
After all, this is open source :-)
Leif
I think a modification should be done around here to return busy if just
one channel was busy (only enabled if an option on dial is set)
in asterisk-1.6.0.15/apps/app_dial.c, line 610
Is somebody willing to try?
while (*to !peer) {
struct chanlist *o;
int pos = 0; /* how many channels do we handle */
int numlines = prestart;
struct ast_channel *winner;
struct ast_channel *watchers[AST_MAX_WATCHERS];
watchers[pos++] = in;
for (o = outgoing; o; o = o-next) {
/* Keep track of important channels */
if (ast_test_flag64(o, DIAL_STILLGOING) o-chan)
watchers[pos++] = o-chan;
numlines++;
}
if (pos == 1) { /* only the input channel is available */
if (numlines == (num.busy + num.congestion + num.nochan)) {
ast_verb(2, Everyone is busy/congested at this time
(%d:%d/%d/%d)\n, numlines, num.busy, num.congestion, num.nochan);
if (num.busy)
strcpy(pa-status, BUSY);
else if (num.congestion)
strcpy(pa-status, CONGESTION);
else if (num.nochan)
strcpy(pa-status, CHANUNAVAIL);
} else {
ast_verb(3, No one is available to answer at this time
(%d:%d/%d/%d)\n, numlines, num.busy, num.congestion, num.nochan);
}
*to = 0;
return NULL;
}
Preferably, either the dialcommand should be preceeded with a
ChanIsAvail on the sip first, as there is no need to place a toll-call
to the cell if the sip is busy. Or the dialcommand itself should have an
option to delay one or more of the calls in the dialstring
(Dial(Technology/resource[Tech2/resource2...]). But this would probably
be too messy...
Leif
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Re: [asterisk-users] ASTERISK and SNMP
Hi michal, see below my ifconfig result : eth0 Link encap:Ethernet HWaddr 00:09:6B:A3:74:4B inet addr:192.168.2.13 Bcast:192.168.2.255 Mask:255.255.255.0 inet6 addr: fe80::209:6bff:fea3:744b/64 Scope:Link UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1 RX packets:594773 errors:0 dropped:0 overruns:0 frame:0 TX packets:535227 errors:0 dropped:0 overruns:0 carrier:0 collisions:0 txqueuelen:1000 RX bytes:67559645 (64.4 MiB) TX bytes:217992551 (207.8 MiB) Interrupt:185 loLink encap:Local Loopback inet addr:127.0.0.1 Mask:255.0.0.0 inet6 addr: ::1/128 Scope:Host UP LOOPBACK RUNNING MTU:16436 Metric:1 RX packets:1187110 errors:0 dropped:0 overruns:0 frame:0 TX packets:1187110 errors:0 dropped:0 overruns:0 carrier:0 collisions:0 txqueuelen:0 RX bytes:191364585 (182.4 MiB) TX bytes:191364585 (182.4 MiB) James, thanks a lot for your advice, I will have a look to the net SNMP daemon wich is map to a AMI command regards Mickael 2009/11/28 Mr. James W. Laferriere bab...@baby-dragons.com Hello Mickael , On Fri, 27 Nov 2009, mickael ropars wrote: Michal, in the IF-MIB you only have 4 interfaces lo,eth0,eth1,sit0 l0 which is the loopback interface eth0, eth1 : ethernet interface sit0 : use for PTP tunneling (use for IPv6) so no information on the digium interface. my IF MIB has also those interfaces I found one the solution to get status of the cards, and all snmp data. the solution is argus : http://argus.tcp4me.com/ with this tools you can have a complete view of your system. regards Mickael While Argus is quite good at monitoring systems and is rather easy to manage . In the case of Asterisk monitoring it uses the Asterisk Management Interface (ie: AMI) not snmp . I was( and still am) hoping that the same information available to the administrator thru the AMI can/will be made available thru snmp polling traps . It should not be too difficult to make net-snmp's daemon make those connections to AMI locally on the asterisk server then report that data back to the snmp client . But everytime I've tried to expand snmpd's functionality I've hit nothing but failures . Twyl , JimL 2009/11/27 michal kalinowski michal.kalinow...@interia.pl Check this command snmpwalk -c your_community -v 1 localhost interfaces in my system it's looks like that: IF-MIB::ifNumber.0 = INTEGER: 4 IF-MIB::ifIndex.1 = INTEGER: 1 IF-MIB::ifIndex.2 = INTEGER: 2 IF-MIB::ifIndex.3 = INTEGER: 3 IF-MIB::ifIndex.4 = INTEGER: 4 IF-MIB::ifDescr.1 = STRING: lo IF-MIB::ifDescr.2 = STRING: eth0 IF-MIB::ifDescr.3 = STRING: eth1 IF-MIB::ifDescr.4 = STRING: sit0 IF-MIB::ifType.1 = INTEGER: softwareLoopback(24) IF-MIB::ifType.2 = INTEGER: ethernetCsmacd(6) IF-MIB::ifType.3 = INTEGER: ethernetCsmacd(6) IF-MIB::ifType.4 = INTEGER: tunnel(131) IF-MIB::ifMtu.1 = INTEGER: 16436 IF-MIB::ifMtu.2 = INTEGER: 1500 IF-MIB::ifMtu.3 = INTEGER: 1500 IF-MIB::ifMtu.4 = INTEGER: 1480 IF-MIB::ifSpeed.1 = Gauge32: 1000 IF-MIB::ifSpeed.2 = Gauge32: 1000 IF-MIB::ifSpeed.3 = Gauge32: 10 IF-MIB::ifSpeed.4 = Gauge32: 0 IF-MIB::ifPhysAddress.1 = STRING: IF-MIB::ifPhysAddress.2 = STRING: 0:14:5e:32:15:70 IF-MIB::ifPhysAddress.3 = STRING: 0:14:5e:32:15:71 IF-MIB::ifPhysAddress.4 = STRING: IF-MIB::ifAdminStatus.1 = INTEGER: up(1) IF-MIB::ifAdminStatus.2 = INTEGER: down(2) IF-MIB::ifAdminStatus.3 = INTEGER: up(1) IF-MIB::ifAdminStatus.4 = INTEGER: down(2) IF-MIB::ifOperStatus.1 = INTEGER: up(1) IF-MIB::ifOperStatus.2 = INTEGER: down(2) IF-MIB::ifOperStatus.3 = INTEGER: up(1) IF-MIB::ifOperStatus.4 = INTEGER: down(2) IF-MIB::ifLastChange.1 = Timeticks: (0) 0:00:00.00 IF-MIB::ifLastChange.2 = Timeticks: (0) 0:00:00.00 IF-MIB::ifLastChange.3 = Timeticks: (0) 0:00:00.00 IF-MIB::ifLastChange.4 = Timeticks: (0) 0:00:00.00 IF-MIB::ifInOctets.1 = Counter32: 37919437 IF-MIB::ifInOctets.2 = Counter32: 0 IF-MIB::ifInOctets.3 = Counter32: 1491657594 IF-MIB::ifInOctets.4 = Counter32: 0 IF-MIB::ifInUcastPkts.1 = Counter32: 335932 IF-MIB::ifInUcastPkts.2 = Counter32: 0 IF-MIB::ifInUcastPkts.3 = Counter32: 162961409 IF-MIB::ifInUcastPkts.4 = Counter32: 0 IF-MIB::ifInNUcastPkts.1 = Counter32: 0 IF-MIB::ifInNUcastPkts.2 = Counter32: 0 IF-MIB::ifInNUcastPkts.3 = Counter32: 131166 IF-MIB::ifInNUcastPkts.4 = Counter32: 0 IF-MIB::ifInDiscards.1 = Counter32: 0 IF-MIB::ifInDiscards.2 = Counter32: 0 IF-MIB::ifInDiscards.3 = Counter32: 0 IF-MIB::ifInDiscards.4 = Counter32: 0 IF-MIB::ifInErrors.1 = Counter32: 0 IF-MIB::ifInErrors.2 = Counter32: 0 IF-MIB::ifInErrors.3 = Counter32: 0 IF-MIB::ifInErrors.4 = Counter32: 0 IF-MIB::ifInUnknownProtos.1 = Counter32: 0 IF-MIB::ifInUnknownProtos.2 =
Re: [asterisk-users] Prevent Dial if any extension is busy
Leif Neland le...@neland.dk writes: I think a modification should be done around here to return busy if just one channel was busy (only enabled if an option on dial is set) in asterisk-1.6.0.15/apps/app_dial.c, line 610 That is doable, but it can result in a bad experience for the caller. The Dial() is likely to indicate progress to the caller, which means that the caller will hear the familiar dialing tone (By the way, is there a dictionary of the names for the various telecoms tunes?). Right afterwards they will hear the busy tone, as if the callee rejected the call. It is best not to send a busy tone once you have indicated that the call is on the way to being connected -- unless you're trying to get rid of a telemarketer. /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Warning: __ast_register_translator: plc_samples 160 format f/__ast_string_field_init: trying to reset empty pool
In a (futile?) attempt to get rid of warnings, I have this: [Nov 30 10:39:49] NOTICE[68467]: loader.c:937 load_modules: 149 modules will be loaded. [Nov 30 10:39:49] WARNING[68467]: utils.c:1427 __ast_string_field_init: trying to reset empty pool (5 times more) SIP channel loading... (5 lines of AEL loading) [Nov 30 10:39:49] NOTICE[68467]: pbx_ael.c:149 pbx_load_module: AEL load process: verified config file name '/usr/local/etc/asterisk/extensions.ael'. [Nov 30 10:39:49] WARNING[68467]: translate.c:641 __ast_register_translator: plc_samples 160 format f [Nov 30 10:39:49] NOTICE[68467]: config.c:1923 ast_config_engine_register: Registered Config Engine curl Googling these two warnings give nothing usable (for me...) Leif ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ISDN30 Timing Sources (Jon Morgan)
Thanks for that Russell. Seems the only difference we have is that you have a cable 133 feet. :-) I'm baffled as to why we have these issue now. It's been working fine for years but just started getting all these pops, clicks and calls cutting out recently. Cheers, Jon. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Russell Brown Sent: 27 November 2009 09:33 To: asterisk-users@lists.digium.com Subject: [asterisk-users] ISDN30 Timing Sources (Jon Morgan) Quoth Jon Morgan jon.mor...@motors.co.uk We have a 2 port Digium TE220P card, one span is configured to connect to our ISDN30 provider (British Telecom), the other span connects to our internal PBX. Here's the zaptel.conf snip: span=1,1,0,ccs,hdb3,crc4 bchan=1-15 dchan=16 bchan=17-31 span=2,0,0,ccs,hdb3,crc4 bchan=32-46 dchan=47 bchan=48-62 FWIW, I (also BT ISDN30 on span 1 with a PBX on the second port of a TE205P) have the following zaptel.conf. span=1,1,1,ccs,hdb3,crc4 bchan=1-15 dchan=16 bchan=17-31 span=2,0,1,ccs,hdb3,crc4 bchan=32-46 dchan=47 bchan=48-62 I do all my call recording in asterisk so can't comment on that but the PBX users are not complaining about the quality. -- Regards, Russell | Russell Brown | MAIL: russ...@lls.com PHONE: 01780 471800 | | Lady Lodge Systems | WWW Work: http://www.lls.com | | Peterborough, England | WWW Play: http://www.ruffle.me.uk | ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Jon Morgan Direct Dial: Mobile: Motors.co.uk Ltd is registered in England and Wales with Company No. 5975777. VAT registration GB 243 5711 74 Registered address: Northcliffe House, 2 Derry Street, London W8 5TT. Motors.co.uk Ltd is a Daily Mail and General Trust plc company. This message and any attachments are intended for the named addressee only. It contains information which may be confidential, legally privileged or otherwise protected by law. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachments from your system. If you are not the intended recipient you must not copy this message or attachments or disclose the contents to any other person. Emails are not secure and may contain viruses. Motors.co.uk reserves the right to monitor all e-mail communications. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Max how many users in sip.conf
Tilghman Lesher wrote: On Sunday 29 November 2009 17:03:04 Leif Neland wrote: mtha...@gmail.com skrev: Anyone know how many users i can record in sip.conf. (NO..NO i am not discussing the simultaneous sip calls). I tried with 50k users in sip.conf, but the sip module didn't reload. tried with few hundred of users and it works. any idea what is the limit in sip.conf Try a binary search in 15 tries you have the exact value. Start with 32768 entries. If it works, add 32768/2 =16384. If not, subtract 16384, giving 16384. Continue, adding/subtracting 8192,4096.2048,1024,512,256,128,64,32,16.8,4,2,1 There is no maximum. However, if you have a typo in there somewhere, the entire file will fail to load. Not sure that is correct. I added some garbage, and while I got warnings, the rest of the users loaded correctly WARNING[79673]: config.c:1124 process_text_line: No '=' (equal sign) in line 104 of /usr/local/etc/asterisk/sip.conf WARNING[79673]: chan_sip.c:22771 reload_config: Unknown type 'friendly' for '36949608' in sip.conf Line numbers on all errors would be nice, but perhaps the second line is syntactly correct, and is parsed by config.c, but when chan_sip tries to use it, and see it gives no meaning, it does not know where the info came from in the first place. Leif ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No application 'ReceiveFAX' - Solved
Did a recompile of everything, and then it started to work. Must have missed somthing when I did the first compile, or I did something in wrong order. DId a test with a fax machine attached to a POTS interface on an Avaya CM, H.323 trunk to Asterisk. Manage to send from the fax machine to the Asterisk server. Need to work on the T.38 and H.323 in the Asterisk, had to disable T.38 in the Avaya CM to be able to get the fax through. Can keep u posted if u are intrested in how it goes. On Mon, 30 Nov 2009 09:32:14 +0100, Magnus Benngård wrote: Hi! Have probably not understand how fax is working in Asterisk 1.6. I did install: ptlib-v1_12_0 h323plus-v1_19_7 dahdi-linux-complete-2.2.0.2+2.2.0 spandsp-0.0.5 asterisk-1.6.2 asterisk-addons-1.6.2 make menuselect in asterisk-1.6.2 source directory shows: [*] app_fax But core show applications doesnt show me any fax applications and when I try to receive a fax: exten = 960,1,Answer() exten = 960,2,Wait(3) exten = 960,3,ReceiveFAX(/var/spool/asterisk/tmp/fax.tiff) [Nov 30 09:29:18] WARNING[19893]: pbx.c:3677 pbx_extension_helper: No application 'ReceiveFAX' for extension (inputinterior.se, 960, 3) Can any guru guide me what I am doing wrong? Best regards, MAGNUS BENNGRD Direktnr 031-799 89 75 Fältspatsgatan 2 421 30 Västra Frölunda Tel. 031-799 89 00 Fax 031-799 89 01 www.inputinterior.se [1] Links: -- [1] http://www.inputinterior.se inline: 7___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No application 'ReceiveFAX'
Run a module load app_fax.so on asterisk console and see what happens. Regards 2009/11/30 Magnus Benngård magnu...@inputinterior.se Hi! Have probably not understand how fax is working in Asterisk 1.6. I did install: ptlib-v1_12_0 h323plus-v1_19_7 dahdi-linux-complete-2.2.0.2+2.2.0 spandsp-0.0.5 asterisk-1.6.2 asterisk-addons-1.6.2 make menuselect in asterisk-1.6.2 source directory shows: [*] app_fax But core show applications doesnt show me any fax applications and when I try to receive a fax: exten = 960,1,Answer() exten = 960,2,Wait(3) exten = 960,3,ReceiveFAX(/var/spool/asterisk/tmp/fax.tiff) [Nov 30 09:29:18] WARNING[19893]: pbx.c:3677 pbx_extension_helper: No application 'ReceiveFAX' for extension (inputinterior.se, 960, 3) Can any guru guide me what I am doing wrong? Best regards, *Magnus Benngård* Direktnr 031-799 89 75 [image: mailfot] Fältspatsgatan 2 421 30 Västra Frölunda Tel. 031-799 89 00 Fax 031-799 89 01 www.inputinterior.se ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Eduardo Vieira 7___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] UniMRCP Integrated Asterisk Deployment
I'd like to announce the release of an open source connector bridge for Asterisk and UniMRCP. The connector bridge is an implementation of Asterisk's Generic Speech API using UniMRCP client stack. This module allows Asterisk to connect to MRCPv2 or MRCPv1 compliant servers for speech recognition. It also allows to offload Asterisk using client/server architecture MRCP provides. Moreover, all the speech applications, which are written on top of Generic Speech API now can be easily used via MRCP too. The complete package, which includes everything required for UniMRCP integrated Asterisk deployment, is now available for download http://unimrcp.googlecode.com/files/uni-ast-package.tar.gz The open development model with community driven initiatives is what I started UniMRCP with and entered the open source world for. The integration with powerful telephony platforms such as Asterisk would allow the project to further evolve and accelerate. I have been asked many times for such an integration. Now it's your turn to give it a try, provide feedback and help carefully test and evolve the solution. Thanks for using UniMRCP, -- Arsen Chaloyan The author of UniMRCP http://www.unimrcp.org___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TE412P with zaptel
Hi Kevin, Thanks for the reply. So purchasing TE412P with VPMOCT128 echo-cancellation module is not going to effect the current process? It will work with asterisk-1.2.17, zaptel-1.2.17.1. Correct? Regards, Kurian Thayil. On Sat, Nov 28, 2009 at 8:14 PM, Kevin P. Fleming kpflem...@digium.comwrote: Kurian Thayil wrote: I understand that new TE412P interface card comes with VPMOCT128 echo-cancellation module. I already have a server installed with asterisk-1.2.17, zaptel-1.2.17.1 in production with TE412P card installed already and having VPM450 echo cancellation module and it works. I need to purchase a backup card. So, I just need to know whether zaptel driver which is installed in my server is compatible with new TE412P having VPMOCT128 echo cancellation module. Thanks in advance. The VPM450M and VPMOCT128 are the same module; the part number was changed some time after it was first released. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] can't hear anything at incoming calls
If all signs point to mis-configuration of your firewall, why not prove them wrong (while in the process getting more details) just add wireshark to the mix. You can then watch the traffic and be able to quickly identify if any is being lost due to blocked ingress/egress ports. DJ On Sat, Nov 28, 2009 at 8:49 AM, Michael Herrmann mich...@intakt-musikinstitut.de wrote: Hi out there, I think i've everything set up properly, outbound calls are working fine, but at incoming calls I can't hear anything, but the other one is able to hear me perfectly. I'm using an asterisk 1.6.1.10 in my internal network in a NAT, connected to my sip-provider using a trunk. Firewall settings on the router are: forward UDP port 5060,5004,1-2 to asterisk server Firewall on asterisk server is switched off (for better testing) [sip.conf] [general] port=5060 bindaddr=0.0.0.0 language = de externhost=intakt-musik.dyndns.org externip=intakt-musik.dyndns.org localnet=192.168.0.0/255.255.255.0 nat=yes http://192.168.0.0/255.255.255.0%0Anat=yes register = USER:p...@sipconnect.sipgate.de/USER [sipconnect.sipgate.de] type = friend host = sipconnect.sipgate.de outboundproxy = sipconnect.live.sipgate.de port = 5060 username = USER fromuser = USER fromdomain = sipconnect.sipgate.de secret = PASS dtmfmode = rfc2833 insecure = port,invite ;insecure = very canreinvite = no registertimeout = 600 disallow = all allow = alaw allow = ulaw context = sipconnect.sipgate.de Auntie google is not very helpful for me. They all say, it looks like a firewall problem on the router. But I'm sure this is set up correctly. Any ideas? -- Michael Herrmann ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ((uint64_t)1 35) is zero! was: Prevent Dial if any extension is busy
Leif Neland wrote:
I think a modification should be done around here to return busy if
just one channel was busy (only enabled if an option on dial is set)
in asterisk-1.6.0.15/apps/app_dial.c, line 610
Is somebody willing to try?
while (*to !peer) {
struct chanlist *o;
int pos = 0; /* how many channels do we handle */
int numlines = prestart;
struct ast_channel *winner;
struct ast_channel *watchers[AST_MAX_WATCHERS];
watchers[pos++] = in;
for (o = outgoing; o; o = o-next) {
/* Keep track of important channels */
if (ast_test_flag64(o, DIAL_STILLGOING) o-chan)
watchers[pos++] = o-chan;
numlines++;
}
Adding this here
if (num.busy) {
strcpy(pa-status, BUSY);
*to = 0;
return NULL;
}
Seems to work
if (pos == 1) { /* only the input channel is available */
if (numlines == (num.busy + num.congestion + num.nochan)) {
ast_verb(2, Everyone is busy/congested at this time
(%d:%d/%d/%d)\n, numlines, num.busy, num.congestion, num.nochan);
if (num.busy)
strcpy(pa-status, BUSY);
However, I tried adding an option OPT_SINGLE_BUSY after these:
#define DIAL_STILLGOING (1 31)
#define DIAL_NOFORWARDHTML ((uint64_t)1 32) /* flags are now 64
bits, so keep it up! */
#define OPT_CANCEL_ELSEWHERE ((uint64_t)1 33)
#define OPT_PEER_H ((uint64_t)1 34)
#define OPT_SINGLE_BUSY ((uint64_t)1 35)
but all these constants have the value zero!
I'm compiling on FreeBSD, asterisk seems to work anyway...
Whats going on?
Leif
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Re: [asterisk-users] Max how many users in sip.conf
It is limited by the amount of memory available to your computer. Each user takes up a chunk of available memory. Let's say for arguments sake that the amount is 4kb (using top might give you a better idea of the real usage and what you're starting with). 50K users at 4kb apiece would use 200mb and most smaller servers start with 2gb or (much) less available. If you had 1gb of memory, a 200mb load with everything else would be pretty taxing. Hope this is helpful. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of mtha...@gmail.com Sent: Saturday, November 28, 2009 5:53 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Max how many users in sip.conf Anyone know how many users i can record in sip.conf. (NO..NO i am not discussing the simultaneous sip calls). I tried with 50k users in sip.conf, but the sip module didn't reload. tried with few hundred of users and it works. any idea what is the limit in sip.conf regards Mike ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Max how many users in sip.conf
snip If you had 1gb of memory, a 200mb load with everything else would be pretty taxing. Hope this is helpful. /snip What distro are you using?? If linux is using 800Mb of memory in an idle state for anything other than file system caching, there's a problem... -Dave From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of mtha...@gmail.com Sent: Saturday, November 28, 2009 5:53 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Max how many users in sip.conf Anyone know how many users i can record in sip.conf. (NO..NO i am not discussing the simultaneous sip calls). I tried with 50k users in sip.conf, but the sip module didn't reload. tried with few hundred of users and it works. any idea what is the limit in sip.conf regards Mike ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ((uint64_t)1 35) is zero! was: Prevent Dial if any extension is busy
Leif Neland wrote: #define OPT_PEER_H ((uint64_t)1 34) #define OPT_SINGLE_BUSY ((uint64_t)1 35) but all these constants have the value zero! I'm compiling on FreeBSD, asterisk seems to work anyway... Whats going on? doh... 64 bits doesn't fit in %d %llu works better. Leif ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Audio issue in skype for asterisk
Hi, we have a similar problem. When we try to make two skype-calls at a time, only one of them has working audio. For this to happen, both calls must be ringing at the same time. Does anyone know how to fix this? Best regards, Marcus Hunger On Thu, Oct 22, 2009 at 10:45 AM, Samir Doshi smrdo...@gmail.com wrote: Hi, I am facing audio issue in my skype for asterisk setup. *Flow of the call is like this.* e.g. Skype users : test2 Sip users: 1001 1002 -- test2 This both sip users 1001 and 1002 are register in same asterisk. And also test2 skype user is register in same asterisk. Now 1001 is dialing skype user test2 (skypeout). So, test2 is getting call. But as test2 skype user is register in our asterisk, our asterisk is getting that call (skypein). And test2 is mapped with 1002 user. So when test2 user call comes to asterisk our asterisk is dialing SIP/1002. And 1002 is getting calls. But when 1001 and 1002 user is connecting they are not getting audio. But this is working fine for only skypout and skypein. But when call come back to asterisk audio issue is coming. I have checked rtp debug, But getting proper packages in rtp debug. I am attaching image of call flow. [image: ?ui=2view=attth=1247ba299ad2be9dattid=0.1disp=attdrealattid=ii_1247ba299ad2be9dzw] Please help me to fix the issue. -- Thanks, Samir Doshi ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Dipl.-Inf. (FH) Marcus Hunger - hun...@sipgate.de Telefon: +49 (0)211-63 55 55-61 Telefax: +49 (0)211-63 55 55-22 sipgate GmbH - Gladbacher Str. 74 - 40219 Düsseldorf HRB Düsseldorf 39841 - Geschäftsführer: Thilo Salmon, Tim Mois Steuernummer: 106 / 5724 / 7147, Umsatzsteuer-ID: DE219349391 www.sipgate.de - www.sipgate.at - www.sipgate.co.uk skypeforasterisk.jpeg___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Max how many users in sip.conf
I'm running CENTOS 5.3 with apache 2, asterisk 1.4.26.2, mysql 5 and php 5.2.11. top shows 928mb out of 1035mb in use with idle asterisk and 17 users. There could be a problem, but I'm relatively new to CENTOS, so any suggestions would be happy. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Gibbons Sent: Monday, November 30, 2009 8:27 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Max how many users in sip.conf snip If you had 1gb of memory, a 200mb load with everything else would be pretty taxing. Hope this is helpful. /snip What distro are you using?? If linux is using 800Mb of memory in an idle state for anything other than file system caching, there's a problem. -Dave _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of mtha...@gmail.com Sent: Saturday, November 28, 2009 5:53 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Max how many users in sip.conf Anyone know how many users i can record in sip.conf. (NO..NO i am not discussing the simultaneous sip calls). I tried with 50k users in sip.conf, but the sip module didn't reload. tried with few hundred of users and it works. any idea what is the limit in sip.conf regards Mike ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Free Polycom Provisioning Tool
In 2007, I released a Polycom Provisioning Tool. I retired the package earlier this year, and have had so many requests for it, I have revived the concept, new, improved, and still FREE. Any chance of you releasing the source? The asterisk GUI does Polycom phone provisioning, and that source is definitely available. - Noah ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Max how many users in sip.conf
I’m running CENTOS 5.3 with apache 2, asterisk 1.4.26.2, mysql 5 and php 5.2.11. top shows 928mb out of 1035mb in use with idle asterisk and 17 users. There could be a problem, but I’m relatively new to CENTOS, so any suggestions would be happy. I use CentOS for asterisk boxen, too, and my first task after installing the OS is always to use chkconfig to disable the many totally unnecessary processes that are on by default. I can usually get it down to around 400MB - 600MB used, including asterisk (mostly small offices with less than 20 users). I never use mysql or any other dbms, though. - Noah ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom 500 format file system on every reboot
I have one client that is telling me that their Polycom 500's format the file system every time they reboot, and also that they are unable to make changes locally on the phone itself, only via the config files. If the config file is not available when they try to boot the phone, then they receive an error about not being able to find the config file and then the phone will not boot up. Has anyone seen anything like this before? -- Thanks, --Warren Selby http://www.selbytech.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom 500 format file system on every reboot
Hi Warren - I have one client that is telling me that their Polycom 500's format the file system every time they reboot, and also that they are unable to make changes locally on the phone itself, only via the config files. If the config file is not available when they try to boot the phone, then they receive an error about not being able to find the config file and then the phone will not boot up. Has anyone seen anything like this before? Yes, I've seen this on a number of 500's running very recent versions of the firmware and bootrom. It only seems to affect a small number of the 500's I've worked with, though. Many of them are fine. It hasn't been a big deal for anybody as I always do provisioning through FTP, and the phones rarely need to be rebooted. I'm trying desperately to get rid of all the 500's I have out in service. Just so many bugs on them. - Noah ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Please some enlightment on ENUM !!
Norbert Zawodsky wrote: But then you create phonenumbers in enum, which doesn't exist as pstn-numbers. Not the idea behind enum. On the other hand, if you owned 10 or 100 pstn-numbers in series, you could get the last one or two digits delegated to your dns-server. Leif ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Why do I create numbers in enum which doesn't exist as pstn ? A simple example: My pstn number is +43-1-1234567. Everybody around the world can call me using this number. Lets say, I have 3 extensions: 0=reception, 10=secretary, 20=boss. If someone calls ENUMLOOKUP(+4311234567) he will get a uri sip:0...@ip.of.my.asterisk ENUMLOOKUP(+43112345670) he will get a uri sip:0...@ip.of.my.asterisk ENUMLOOKUP(+431123456710) he will get a uri sip:s...@ip.of.my.asterisk or sip:1...@ip.of.my.asterisk (which ever you prefer) ENUMLOOKUP(+431123456720) he will get a uri sip:b...@ip.of.my.asterisk or sip:2...@ip.of.my.asterisk All this numbers exist because they connect to different persons. Why shouldn't that be the idea behind enum? Norbert ENUM is, quite literally, E164 Number Mapping (that's what it stands for). If you're mapping numbers which are invalid E164 numbers (i.e. in your scenario in which you're taking an E164 number and attaching digits to it), you're violating the ENUM idea for the sake of convenience. You're also making the somewhat unfounded assumption that there will never be an actual number issued (to someone else) with those extra digits. Right NOW, there may be a convention that says that you can only have 10 digits in your country's phone numbers, but that could conceivably change at some future date, and then you'd be mapping numbers that belong to someone else to your own services. The only VALID way to assign ENUM numbers is to assign numbers you actually own. Not numbers you own with additional digits. Not numbers you own with extentions tacked on. Not numbers that are similar to ones you own. But ONLY ones you own. In this case, you own +4321234567, and only THAT number should be allowed to be registered as an ENUM number. Unless you, for instance, also own +4321234568 and +4321234569 or some such... at which time you would certainly be able to register those numbers and point them to your PBX. What you're suggesting, though, violates the ENUM standard... and should not be allowed. N. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Parsing custom SIP headers
Philipp Kempgen wrote: Just to be sure: Is there a dialplan function in Asterisk that parses custom name-addr-style SIP headers for me? Try this: https://issues.asterisk.org/view.php?id=16268 Leif Madsen. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Parsing custom SIP headers
Leif Madsen schrieb: Philipp Kempgen wrote: Just to be sure: Is there a dialplan function in Asterisk that parses custom name-addr-style SIP headers for me? Try this: https://issues.asterisk.org/view.php?id=16268 Thanks but I don't see the connection. Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Please some enlightment on ENUM !!
SIP schrieb: ENUM is, quite literally, E164 Number Mapping (that's what it stands for). If you're mapping numbers which are invalid E164 numbers (i.e. in your scenario in which you're taking an E164 number and attaching digits to it), you're violating the ENUM idea for the sake of convenience. You're also making the somewhat unfounded assumption that there will never be an actual number issued (to someone else) with those extra digits. Right NOW, there may be a convention that says that you can only have 10 digits in your country's phone numbers, but that could conceivably change at some future date, and then you'd be mapping numbers that belong to someone else to your own services. The only VALID way to assign ENUM numbers is to assign numbers you actually own. Not numbers you own with additional digits. Not numbers you own with extentions tacked on. Not numbers that are similar to ones you own. But ONLY ones you own. In this case, you own +4321234567, and only THAT number should be allowed to be registered as an ENUM number. Unless you, for instance, also own +4321234568 and +4321234569 or some such... at which time you would certainly be able to register those numbers and point them to your PBX. What you're suggesting, though, violates the ENUM standard... and should not be allowed. N. Sorry N. ! But - at least here in Austria - it is definitely *no* assumption that my number with some extra digits can not be issued to someone else. The number +43-1-3207978 is my telephone number. I own it as long as I pay for it. And with extra digits behind it I can do whatever I like. I can create any extension - physical or virtual. I can attach a phone to extension 12, attach a virtual fax server for extension 12 to extension 99912 or could fire up my toaster if I call extension 911. I can invent any numbering scheme for my company. That's a fact! Again - At least here in Austria !! (can't speak for other countries) And why would nic.at (the owner of our .at TLD) offer the possibility to register a e164 domain specific Nameserver to answer subdomain-requests for your number if it would violate ENUM standards? I don't think that they're not knowing what they do It *is* the same as with normal domains. If you rent myhome.org you can create any subdomain like razor.bathroom.myhome.org ;-) And 7.6.5.4.3.2.1.3.4.e164.arpa is my domain (also literally! What does the word domain mean?), and anything below that in the domain tree is under my responsibility. N. (too) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DAHDI - BRI - Astribank
Hello List, it is a very long time since I wrote here It has been still in Zaptel times Today I am run into a related problem: I can't get a DAHDI setup to work 100%. I am configuring an Astribank XR00013 (BRI, two ISDN ports). At some degree the installation (latest DAHDI drivers, Asterisk 1.6.0.18 on Centos 5.x) works. I can get incoming calls to the dialplan context that is setup as target. The call quality is good, etc. The problem is on the other side, the outgoing route. There I get just rejected calls. I am including some information on the setup: dahdi-system.conf ** # Autogenerated by /usr/sbin/dahdi_genconf on Mon Nov 30 18:37:14 2009 # If you edit this file and execute /usr/sbin/dahdi_genconf again, # your manual changes will be LOST. # Dahdi Configuration File # # This file is parsed by the Dahdi Configurator, dahdi_cfg # # Span 1: XBUS-00/XPD-00 Xorcom XPD #00/00: BRI_TE (MASTER) AMI/CCS span=1,1,0,ccs,ami # termtype: te bchan=1-2 hardhdlc=3 echocanceller=mg2,1-2 # Span 2: XBUS-00/XPD-01 Xorcom XPD #00/01: BRI_TE AMI/CCS span=2,2,0,ccs,ami # termtype: te bchan=4-5 hardhdlc=6 echocanceller=mg2,4-5 # Global data loadzone= it defaultzone = it ** dahdi-init.conf ** # # Shell settings for Dahdi initialization scripts. # This replaces the old/per-platform files (/etc/sysconfig/zaptel, # /etc/defaults/zaptel) # # The maximal timeout (seconds) to wait for udevd to finish generating # device nodes after the modules have loaded and before running dahdi_cfg. #DAHDI_DEV_TIMEOUT=40 # Override settings for xpp_fxloader #XPP_FIRMWARE_DIR=/usr/share/dahdi #XPP_HOTPLUG_DISABLED=yes ** chan_dahdi.conf ** ; ; DAHDI telephony ; ; Configuration file [trunkgroups] [channels] language=it ;context=from-pstn ;signalling=fxs_ks ;rxwink=300 ; Atlas seems to use long (250ms) winks ; ; Whether or not to do distinctive ring detection on FXO lines ; ;usedistinctiveringdetection=yes usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no ;echotraining=800 rxgain=0.0 txgain=0.0 group=0 callgroup=1 pickupgroup=1 immediate=no ;faxdetect=both ;faxdetect=incoming ;faxdetect=outgoing faxdetect=no ;Include setup-pstn configs ;internationalprefix = 00 ;nationalprefix = 0 ; Span 1: XBUS-00/XPD-00 Xorcom XPD #00/00: BRI_TE (MASTER) ;group=0,11 group=0 context=from-pstn switchtype = euroisdn signalling = bri_cpe_ptmp channel = 1-2 ;group = 63 ; Span 2: XBUS-00/XPD-01 Xorcom XPD #00/01: BRI_TE ;group=0,12 group=1 context=from-pstn switchtype = euroisdn signalling = bri_cpe_ptmp channel = 4-5 ;group = 63 ;group=1 ** This is what I see on the CLI ** -- Executing [...@macro-outcallsrouter:39] Set(SIP/ 23001-000f, CALLERID(num)=221591030) in new stack -- Executing [...@macro-outcallsrouter:40] NoOp(SIP/ 23001-000f, Dial(DAHDI/g0/191,120)) in new stack -- Executing [...@macro-outcallsrouter:41] Dial(SIP/ 23001-000f, DAHDI/g0/191,120) in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called g0/191 -- DAHDI/1-1 is proceeding passing it to SIP/23001-000f -- Channel 0/1, span 1 got hangup request, cause 63 -- Hungup 'DAHDI/1-1' == Everyone is busy/congested at this time (1:0/0/1) ** And this is from the PRI debug ** -- Making new call for cr 32776 Protocol Discriminator: Q.931 (8) len=32 Call Ref: len= 1 (reference 8/0x8) (Originator) Message type: SETUP (5) [04 03 80 90 a3] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) User information layer 1: A-Law (35) [18 01 81] Channel ID (len= 3) [ Ext: 1 IntID: Implicit Other Spare: 0 Preferred Dchan: 0 ChanSel: B1 channel ] [6c 0b 21 80 32 32 31 35 39 31 30 33 30] Calling Number (len=13) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation permitted, user number not screened (0) '' ] [70 04 a1 31 39 31] Called Number (len= 6) [ Ext: 1 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '191' ] [a1] Sending Complete (len= 1) q931.c:3134 q931_setup: call 32776 on channel 1 enters state 1 (Call Initiated) Protocol Discriminator: Q.931 (8) len=10 Call Ref: len= 1 (reference 8/0x8) (Terminator) Message type: CALL PROCEEDING (2) [18 01 89] Channel ID (len= 3) [ Ext: 1 IntID: Implicit Other Spare: 0 Exclusive Dchan: 0 ChanSel: B1 channel ] [27 01 e8] Notification indicator (len= 3): Ext: 1 Diversion activated (DSS1)
[asterisk-users] Asterisk 1.2.37, 1.4.27.1, 1.6.0.19, and 1.6.1.11 Now Available
The Asterisk Development Team has announced the release of Asterisk 1.2.37, 1.4.27.1, 1.6.0.19, and 1.6.1.11. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ These releases have been created in response to a SIP remote crash vulnerability. Additionally, Asterisk versions 1.4.27.1, 1.6.0.19, and 1.6.1.11 also contain an SDP regression fix as described in issue #16268. Asterisk 1.6.0.19, and 1.6.1.11 contain an additional SDP regression fix as described by issue #16238. Information about the SDP issues can be found at https://issues.asterisk.org/view.php?id=16268 and https://issues.asterisk.org/view.php?id=16238 For more information about the details of this vulnerability, please read the security advisory AST-2009-010, which was released at the same time as this announcement. The security advisory is available at http://downloads.asterisk.org/pub/security/AST-2009-010.pdf For a full list of changes in the current releases, please see the ChangeLogs: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.2.37 http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.4.27.1 http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.0.19 http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.1.11 Thank you for your continued support of Asterisk! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AST-2009-010: RTP Remote Crash Vulnerability
Asterisk Project Security Advisory - AST-2009-010 ++ | Product| Asterisk| |--+-| | Summary| RTP Remote Crash Vulnerability | |--+-| | Nature of Advisory | Denial of Service | |--+-| |Susceptibility| Remote unauthenticated sessions | |--+-| | Severity | Critical| |--+-| |Exploits Known| No | |--+-| | Reported On | November 13, 2009 | |--+-| | Reported By | issues.asterisk.org user amorsen| |--+-| | Posted On | November 30, 2009 | |--+-| | Last Updated On| November 30, 2009 | |--+-| | Advisory Contact | David Vossel dvossel AT digium DOT com | |--+-| | CVE Name | CVE-2009-4055 | ++ ++ | Description | An attacker sending a valid RTP comfort noise payload| | | containing a data length of 24 bytes or greater can | | | remotely crash Asterisk. | ++ ++ | Resolution | Upgrade to one of the versions of Asterisk listed in the | || Corrected In section, or apply a patch specified in the | || Patches section.| ++ ++ | Affected Versions| || | Product | Release Series || |--++| | Asterisk Open Source | 1.2.x | All versions | |--++| | Asterisk Open Source | 1.4.x | All versions | |--++| | Asterisk Open Source | 1.6.x | All versions | |--++| |Asterisk Business Edition | B.x.x | All versions | |--++| |Asterisk Business Edition | C.x.x | All versions | |--++| |s800i (Asterisk Appliance)| 1.3.x | All versions | ++ ++ | Corrected In | || | Product | Release | |-+--| |Asterisk Open Source | 1.2.37 | |-+--| |Asterisk Open Source | 1.4.27.1 | |-+--| |Asterisk Open Source | 1.6.0.19 | |-+--| |
[asterisk-users] Asterisk and XMPP Jingle : testers needed
Dear community members,
I'm happy to announce that we now have code that allows you to use
your XMPP (Jabber) client like a softphone to place SIP or PSTN (or
whatever channel Asterisk supports) calls.
The XMPP clients that support Jingle that I and others have tested are :
- Pidgin (Linux, Ubuntu 9.10), version 2.6.2 : OK
- Empathy (Linux, Ubuntu 9.10), version 2.28.1.1 : OK
- Psi (Windows XP), version 0.13 : Call establishes, but no sound
(seems to be a problem with Speex)
For the moment, one can only place calls from the XMPP client to
Asterisk, but soon, you'll be able to receive calls on your XMPP
client too.
Please test the following branch :
http://svn.digium.com/svn/asterisk/team/phsultan/jingle-support
Or visit this ticket : https://issues.asterisk.org/view.php?id=15634
The doc/jabber.txt in the code contains code snippets and
configuration examples. Hereafter is an example of how to place a call
to an Asterisk server through the Jingle channel. The user places a
Jingle call to Asterisk from his XMPP client's UI, which triggers a
chat message being sent back to him, asking him to enter a number to
call. And that's it, Asterisk just relays the call to the configured
destination (here, a registered SIP phone).
context jingle-in {
s = {
Answer();
SendText(Please enter the number you wish to call);
Set(NEWEXTEN=${JABBER_RECEIVE(asterisk-xmpp,${CALLERID(name)})});
SendText(Calling ${NEWEXTEN} ...);
Dial(SIP/${NEWEXTEN);
Hangup();
}
}
Thanks,
Philippe
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[asterisk-users] AGI
I am trying to find an AGI script that runs via PHP and performs the send text application. Does anyone have any tools or scripts set up for this please? If so, kindly send some info or the code that performs this action. Thank you ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI
Hi Thomas, Hope this will be helpful for you: http://www.voip-info.org/wiki/view/Asterisk+AGI+php On Tue, Dec 1, 2009 at 8:46 AM, Thomas Perron thomas.per...@gmail.comwrote: I am trying to find an AGI script that runs via PHP and performs the send text application. Does anyone have any tools or scripts set up for this please? If so, kindly send some info or the code that performs this action. Thank you ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- ===***=== http://www.hidoc.info ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom retrieve call from hold
On Saturday 28 November 2009 06:48:01 pm Darrick Hartman wrote: Mike Diehl wrote: On Saturday 28 November 2009 04:48:13 pm Darrick Hartman wrote: Mike Diehl wrote: On Saturday 28 November 2009 06:54:16 am Kevin P. Fleming wrote: Mike Diehl wrote: On Friday 27 November 2009 11:09:02 am Noah Miller wrote: Hi Mike - I've got a Polycom 501 that's been working with Asterisk for some time. However, I don't seem to be able to put a call on hold and get it back. It goes on hold just fine. But when I press the resume button, nothing happends. Anyone seen this befor? Any ideas on where to start to fix it? Ok, so I've got a very old firmware on my phone that needs to be upgraded. If I remove my provisioning file(s) from my ftp server and rev the firmware, would this completely wipe out any and all prior configuration on the phone? If this would work, I could elliminate about 50K worth of XML from the equation. Then I'd go in and configure the phone via the web. If I have a working phone, then I can conclude that either the upgrade fixed it, or the XML provisioning file I was using was at fault. Does this sound right? Does anyone have a better/different idea? I'd configure the phone using the XML files, but take a look at the method that Karl Fife has documented here: http://www.kfife.com/voip/ Minimal changes are made to files. The base config files are never touched which makes upgrading firmware versions super easy. So I listened to the podcast and read all I could. Turns out that this is a very elegant solution, and it worked. My call hold works. Then I geeked out over the micro-browser and IM capability for about 2 hours. The I got cocky I decided to change my user id from line_1 - line_3 to 0004F211D1D0-1 - 0004F211D1D0-3. I was thinking that this simple change would go quickly and it would make my config much more uniform, as I use the same convention for the Sipura/Linksys/Cisco TA's. The problem is that after changing the the [MACADDRESS].cfg file to contain reg.1.auth.userId=0004F211D1D0-1, the phone is still trying to register with the old credentials. My fsftp log file confirms that the phone is actually downloading the new .cfg file. What am I missing? -- Take care and have fun, Mike Diehl. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Configuration with Sphinx speech engine
hi, i am using asterisk 1.6 and i want to integrate sphinx speech engine with my asterisk, so that i can use the generic speech API provided by asterisk 1.6... Plz help me, how can i do that... any help will be highly appreciated... waiting for your positive response... Thanks Regards,Rizwan HasnaniFinal Year Student - NUCES-FASTEmail id: k060...@nu.edu.pkcell#: 0345-3235008 _ Windows 7: Unclutter your desktop. Learn more. http://www.microsoft.com/windows/windows-7/videos-tours.aspx?h=7secslideid=1media=aero-shake-7secondlistid=1stop=1ocid=PID24727::T:WLMTAGL:ON:WL:en-US:WWL_WIN_7secdemo:122009___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Configuration with Sphinx speech engine
Hello I also tried it in begining but cant give time to it. So no success. you can try this link http://www.voip-info.org/wiki/view/Sphinx http://cmusphinx.sourceforge.net/html/cmusphinx.php http://cmusphinx.sourceforge.net/html/cmusphinx.php hope this helps On Tue, Dec 1, 2009 at 12:16 PM, Rizwan Hasnani rizwanhasn...@hotmail.comwrote: hi, i am using asterisk 1.6 and i want to integrate sphinx speech engine with my asterisk, so that i can use the generic speech API provided by asterisk 1.6... Plz help me, how can i do that... any help will be highly appreciated... waiting for your positive response... Thanks Regards, Rizwan Hasnani Final Year Student - NUCES-FAST Email id: k060...@nu.edu.pk Cell#: 0345-3235008 -- Windows 7: Unclutter your desktop. Learn more.http://www.microsoft.com/windows/windows-7/videos-tours.aspx?h=7secslideid=1media=aero-shake-7secondlistid=1stop=1ocid=PID24727::T:WLMTAGL:ON:WL:en-US:WWL_WIN_7secdemo:122009 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Shakeel Abbas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
