Re: [asterisk-users] chan_sip.c: Failed to parse contact info
Den 19-01-2011 00:19, Nick Ustinov skrev:
Hello!
I have just upgraded to asterisk 1.8.2.1 and see some weird messages
in log when client tries to register:
[2011-01-19 00:52:47] WARNING[25624] chan_sip.c: Failed to parse contact info
[2011-01-19 00:52:50] NOTICE[25624] chan_sip.c: Peer '0010101' is now
UNREACHABLE! Last qualify: 105
[2011-01-19 00:53:03] VERBOSE[25624] chan_sip.c: -- Registered SIP
'0010101' at 78.84.202.65:37891
Could it be because single_binding_found is not initialized to zero?
in chan_sip.c
static enum parse_register_result parse_register_contact(...
...
int wildcard_found = 0;
int single_binding_found;
...
if (!strcasecmp(curi, *)) {
wildcard_found = 1;
} else {
single_binding_found = 1;
}
if (wildcard_found (ast_strlen_zero(expires) ||
expire != 0 || single_binding_found)) {
/* Contact header parameter * detected, so
punt if: Expires header is missing,
* Expires value is not zero, or another
Contact header is present. */
return PARSE_REGISTER_FAILED;
}
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Re: [asterisk-users] asterisk behind nat
Den 02-03-2011 16:12, Jeremy Kister skrev: On 3/2/2011 9:46 AM, Leif Neland wrote: Some of the phones are being disconnected with Asterisk saying no reply to critical packet What kind of phones are they? I might have nothing to do with your network configuration; try adding to sip.conf [general]: session-timers=refuse Did no change. A Budgetone 200 always gets disconnected, appearently not answering this: Retransmitting #5 (no NAT) to 192.168.5.140:5060: SIP/2.0 200 OK^M Via: SIP/2.0/UDP 192.168.5.140:5060;branch=z9hG4bK9fd529935f5b4f0e;received=192.168.5.140^M From: Merethe Neland sip:mere...@arnold.neland.dk;tag=9c97c540dba5aceb^M To: sip:6...@arnold.neland.dk;tag=as4d2cf5b3^M Call-ID: 13bca406eacc2ef8@192.168.5.140^M CSeq: 5145 INVITE^M Server: Asterisk PBX 1.8.2.4^M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH^M Supported: replaces^M Contact: sip:6000@94.18.45.10:5060^M Content-Type: application/sdp^M Content-Length: 204^M ^M v=0^M o=root 1348141594 1348141594 IN IP4 94.18.45.10^M s=Asterisk PBX 1.8.2.4^M c=IN IP4 94.18.45.10^M t=0 0^M m=audio 14144 RTP/AVP 3^M a=rtpmap:3 GSM/8000^M a=silenceSupp:off - - - -^M a=ptime:20^M a=sendrecv^M It is a call from phone 192.168.5.140 to echotest (6000 on 94.18.45.10) The intro from echotest is heard until asterisk disconnects. On a Budgetone 100, it works, getting this line on the console -- Locally bridging SIP/9-0006 and SIP/musimi-0007 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk behind nat
I'm running asterisk on a Freebsd with 2 Nic's. Inside NIC is 192.168.5.x where the phones are. Outside NIC used to be a public IP with the ISP's device set to bridging, but the new WiMAX router only offers me the public ip 94.18.x.x on the outside, and forwarding everything to 192.168.1.50 on the Outside NIC Some of the phones are being disconnected with Asterisk saying no reply to critical packet How is Asterisk supposed to be configured? Currently this: externip = 94.18.x.x ; Address that we're going to put in outbound SIP messages ; if we're behind a NAT localnet = 192.168.5.0 ; Internal NETWORK address localmask = 255.255.255.0 ; Internal netmask ; The externip, localnet and localmask is used ; when registering and communication with other proxies ; that we're registered with tcpbindaddr=0.0.0.0 bindaddr = 0.0.0.0 Leif -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] User on PC?
I'm looking for a way for linux to query a pc if user X is on, and has used the pc recently or the screensaver is not active. If so, I'll route a call for user X to the phone near that PC. Ideas, anyone? Leif -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 911, location
Den 28-01-2010 20:15, Danny Nicholas skrev:
Here's one solution:
- exten = _911,1,Set(IMAT=EXTEN)
- exten = _911,2,Set(IMAT=CUT(IMAT|\/|2)
- exten = _911,3,Dial(DAHDI/1,w911)
- exten = _911,4,Background(emergencyin${IMAT})
Where you would record /var/lib/asterisk/sound/emergencyin100 for extension
100, etc.
I see two problems:
1: Doesn't asterisk see a pots-call as answered as soon as it has
pressed the last digit and therefore will speak into the ring signal?
2: Often callers are answered with an automated message This is 911,
please hold, just to give pranksters/misdiallers a chance to hang up
before disturbing the operator. Unless 911 records the incoming call
right from the start, they will never hear the im-at message. And even
if they do, they have to know the message is there to seek on the recording.
An option of the operator receiving a loop of This is a call from the
Mickey Mouse building room 123, please press * to receive the call
would require the operator to be able to press *, not sure I'd depend
my life on that...
Leif
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Re: [asterisk-users] 911, location
Den 29-01-2010 19:38, Danny Nicholas skrev:
This might help
- exten = _911,1,Set(IMAT=EXTEN)
- exten = _911,2,Set(IMAT=CUT(IMAT|\/|2)
- exten = _911,3,Dial(DAHDI/1,w911)
- exten = _911,4(keepup),Background(emergencyin${IMAT})
- exten = _911,5,wait(10)
- exten = _911,6,Goto(keepup)
This would repeat the message every 10 seconds...
--
This would prevent the caller talking to the 911-operator, wouldn't it?
Leif
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Re: [asterisk-users] 10/100 voip phones and gigabit connection
Jeff LaCoursiere skrev: On Fri, 15 Jan 2010, Hans Witvliet wrote: If you connect your pc with GB-lan card to an dual-ported ip-phone, you and up with an 100Mbps lan connection to your pc. Only way to avoid that, is to insert a cheap second lan-card in your pc, and connect your phone to the second lan, so your pc will act as an switch, instead of your phone... I'm curious - how have you managed to connect a second LAN card and have it bridge your (presumably onboard) ethernet? Does Windows have such capability? But I guess the OP was running XUbuntu, and though relatively complicated I guess you could get it to do that. j On my laptop I just used the controlpanel - network connections , marked wireless and build-in card, rightclicked and selected bridge networks. Then I plugged my ip-phone in the laptop, and my phone was connected via wlan. So at least in Vista it's built in. 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] 10/100 voip phones and gigabit connection
Hans Witvliet skrev: During my last blackout i found out that all but my switches were on the UPS... bummer! Coincidentially, in danish, oops is spelled ups. It also gives funny images when your packages are delivered by a company called Oops... 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] Howto regret blind transfer?
hbk skrev: Hi, Is it possible to regret blind transfer while its ringing (not answered)? Call pickup. If the phone is in your pickup-group. 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] Hint for realtime peers
Tilghman Lesher skrev:
On Saturday 16 January 2010 11:02:52 Deep D wrote:
On Sat, Jan 16, 2010 at 9:21 PM, Tilghman Lesher tles...@digium.com wrote:
On Saturday 16 January 2010 06:04:01 Deep D wrote:
When I create a sip peer in users.conf then a hint is automatically
created for that peer. But when I create a peer in sip.conf or a
realtime peer with the same values then this hint is not created.
Every time I add such peers I have to add a hint in extensions.conf.
Is it possible to have something like exten =
_XXX,hint,SIP/${EXTEN} in extensions.conf so that I don't have to
add hint for each sip peer I create?
Only in 1.6.1 and later. The hints will grow, as phones subscribe to
them, one entry per hint, automatically.
I tried this in asterisk 1.6.1.1 by adding the line exten =
_XXX,hint,SIP/${EXTEN} to the default context, but it did not work.
I gave the following commands through the manager interface
action: extensionstate
exten: 777
and the response was
Response: Success
Message: Extension Status
Exten: 777
Context: default
Hint: SIP/${EXTEN}
Status: 0
I am always getting a Status: 0 for any value of exten. I think the
variable ${EXTEN} is not being evaluated to its value.
That's not a subscription. You must actually get a phone to subscribe to
the hint before it is created.
Might be true for dynamic hints.
But for static hints it's not.
arnold*CLI core show hints
arnold*CLI
-= Registered Asterisk Dial Plan Hints =-
6...@hintcontext : SIP/6
State:IdleWatchers 0
5...@hintcontext : SIP/jesperfon
State:IdleWatchers 1
I have several unused hints.
Leif
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Re: [asterisk-users] 10/100 voip phones and gigabit connection
- Original Message - From: randall To: asterisk-users@lists.digium.com Sent: Friday, January 15, 2010 7:54 AM Subject: [asterisk-users] 10/100 voip phones and gigabit connection hi all, just subscribed to the list and first mail, nice to be here. Hopefully i'm in the right place for this question since i'm planning a little VOIP implementation at the moment and ran in to something while going through the shopping list. i noticed that a lot of VOIP phones have a double network interface allowing you to use only 1 LAN cable for both the phone and your desktop, a really nice feature that can save a lot of cable, but most are 10/100 connections while i have a gigabit network. Off course there are phones available with a Gigabit connection but these are at least 3 to 4 times as expensive. In a pinch, the cheapest 1Gbit switch I could find is 17 Eur with 5 ports. 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] Realtime queue not work
- Original Message -
From: Zhang Shukun
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Friday, January 15, 2010 11:48 AM
Subject: [asterisk-users] Realtime queue not work
hi, all
i try to confiture realtime queue, but not work, details as below:
Insert into queue_table(name)value('95040654321');
INSERT INTO queue_member_table VALUES ('', 'Zhang Shukun',
'95040654321', 'SIP/1001', 2, 1);
INSERT INTO queue_member_table VALUES ('', 'Li Aiwei', '95040654321',
'SIP/1002', 2, 1);
INSERT INTO queue_member_table VALUES ('', 'Zhang Jianming',
'95040654321', 'SIP/1003', 2, 1);
but when i dial 95040654321 and press extension 1. error happens:
-- Executing Queue(SIP/1003-, 950406543211)
[Jan 15 03:18:57] WARNING[16626]: app_queue.c:4223 queue_exec: Unable
to join queue '950406543211'
== Auto fallthrough, channel 'SIP/1003-' status is 'UNKNOWN'
No golden answers, but something to try.
queue names can not be just numbers? I'd try calling the queue q95040654321.
Does show queues show the queue? Don't know if that's supposed to work on
realtime queues.
Leif
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Re: [asterisk-users] 10/100 voip phones and gigabit connection
- Original Message - From: randall To: asterisk-users@lists.digium.com Sent: Friday, January 15, 2010 2:11 PM Subject: Re: [asterisk-users] 10/100 voip phones and gigabit connection On 01/15/2010 02:00 PM, Leif Neland wrote: - Original Message - From: randall To: asterisk-users@lists.digium.com Sent: Friday, January 15, 2010 7:54 AM Subject: [asterisk-users] 10/100 voip phones and gigabit connection list. i noticed that a lot of VOIP phones have a double network interface allowing you to use only 1 LAN cable for both the phone and your desktop, a really nice feature that can save a lot of cable, but most are 10/100 connections while i have a gigabit network. Off course there are phones available with a Gigabit connection but these are at least 3 to 4 times as expensive. In a pinch, the cheapest 1Gbit switch I could find is 17 Eur with 5 ports. Leif its not the network switch that i'm worried about, its the build in switch of the phones with the double network card Sure. My point was just that IF you only got one connection in the wall, its cheaper to get a switch than getting a phone with dual 1Gbit ports. 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] is roundrobin and rrmemory the same meaning?
Zhang Shukun wrote: Thank you! it's very helpful. now i have another question: in asterisk, each agent should login first and then can response to the caller. but i don't want to the login action. i need agent shold response directly without login first. how should i do ? can users in sip.conf to be agents? so it can login persistently on a phone. My phones are listed in queues.conf member = SIP/36949608 member = IAX2/10 member = IAX2/11 -- _ -- 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] Some minor configuration issues with queues
- Original Message - From: jonas kellens To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Monday, January 11, 2010 2:30 PM Subject: Re: [asterisk-users] Some minor configuration issues with queues To answer my own question : I had the following in my dialplan : Queue(VC_support_queue,r) The 'r' option replaces the moh with a dialtone... I have now replaced the 'r' with 'R', so that there is moh and a ringtone when an agent is ringing... (source: voip-info.org) However the caller keeps hearing music on hold in stead of a ring tone when an agent's phone calls. Moh stops when the agents picks up. So I'm not quite there yet. Any feedback is appreciated. Appearently queues are designed to play moh until an agent answers. The caller doesn't need to know if the agent is busy talking on the phone or busy ***ing the secretary :-) You could dial the agent(s) directly with the dial() function, if it returns busy, put the caller into queue. 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
[asterisk-users] Off-line subscribed phone amber on SPA942?
If xlite subscribes on a hint, and the phone is offline, xlite says so (not online) If SPA942 does the same, the led is green for available. The other hints work: blink red for ringing and red for busy. I seem to remember the led once showed amber for subscribed phone offline. The SPA extended function is fnc=blf+sd+cp;su...@my.sip.server Server type is asterisk Share line apperarence is irrelevant, tried both options. Asterisk version 1.6.0.18; latest available in FreeBSD ports. 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
[asterisk-users] {Spam?} MeetMe/Dahdi for FreeBSD
It seems dahdi is needed for meetme, but not available under FreeBSD. So what do I do then? Asterisk has only SIP-channels. ___ -- 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] Looking at Asterisk for 8000sq/ft residential use
Taylor, Jonn skrev: Leif Neland wrote: I can't believe anyone would use RJ-11 any more. You can multi-purpose RJ-45 jacks to work with POTS lines. Run everything down to a central panel and send pots over the jacks that you need to. That way if you decide you need/want to go IP in the future, you're all set. Darrick Better read this before recommending using RJ-45 jacks with RJ-11 line cords. The jacks gets damaged. Manufactures will not warranty them!!! http://www.patentstorm.us/patents/7125288/description.html Jonn You can get or make cords with RJ-45 in one end and RJ-11 in the other. The point is that you should buy the right jacks for the application. Just remember that VIOP is only about 100k of bandwidth per call to a phone. If you are connecting a pc to that phone phone thats different. Jonn Still, if using RJ45 with RJ45-RJ11 cords, one can easily upgrade to voip later without risking the jacks have been damaged. www.lynxbroadband.com offers TV and phone over cat5/6 cables, so to wire phones on two-wire RJ11 is to lock the installation to the technology of the last milleneum. 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] cheap ip phone with auto-answer
Tim Nelson skrev: - Leif Neland le...@neland.dk wrote: I want some cheap ip-phones with auto-answer, to work as paging system at dinnertime. Options, please. Leif I've had great luck using the BT201 phones from Grandstream for this purpose. In fact, this is the only situation where I still use Grandstream handsets. They support auto answer as well as auto answer by call info which allows you to auto answer based upon the SIP header in case you don't want *ALL* calls to be auto answered. I've ordered a couple of BT200, with two ethernet ports, so I don't need a hub on the single ethernet jack. It seems the older BT100 does not have the auto answer by call info with the latest *1.0.8.33*; can it use the firmware for BT20x, 1.2.2.19 http://www.grandstream.com/DOWNLOAD/FIRMWARE/BT200_GXP/Release_BT200_GXP_1.2.2.19.zip ? Or is there hacked versions available? 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] cheap ip phone with auto-answer
Gordon Henderson skrev: On Wed, 30 Dec 2009, Leif Neland wrote: Tim Nelson skrev: - Leif Neland le...@neland.dk wrote: I want some cheap ip-phones with auto-answer, to work as paging system at dinnertime. Options, please. Leif I've had great luck using the BT201 phones from Grandstream for this purpose. In fact, this is the only situation where I still use Grandstream handsets. They support auto answer as well as auto answer by call info which allows you to auto answer based upon the SIP header in case you don't want *ALL* calls to be auto answered. I've ordered a couple of BT200, with two ethernet ports, so I don't need a hub on the single ethernet jack. It seems the older BT100 does not have the auto answer by call info with the latest *1.0.8.33*; can it use the firmware for BT20x, 1.2.2.19 http://www.grandstream.com/DOWNLOAD/FIRMWARE/BT200_GXP/Release_BT200_GXP_1.2.2.19.zip ? Or is there hacked versions available? Not that I'm aware of, but I'm surprised you can still get the BT10x's as I thought they'd been obsoleted... If you are using them, be aware that they have a 10Mb HUB and not a 10/100 Switch that the 20x's have.. I didn't say I could/would get a BT100, I already got two, which don't do auto answer by call info. 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
[asterisk-users] cheap ip phone with auto-answer
I want some cheap ip-phones with auto-answer, to work as paging system at dinnertime. Options, please. 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] Looking at Asterisk for 8000sq/ft residential use
I can't believe anyone would use RJ-11 any more. You can multi-purpose RJ-45 jacks to work with POTS lines. Run everything down to a central panel and send pots over the jacks that you need to. That way if you decide you need/want to go IP in the future, you're all set. Darrick Better read this before recommending using RJ-45 jacks with RJ-11 line cords. The jacks gets damaged. Manufactures will not warranty them!!! http://www.patentstorm.us/patents/7125288/description.html Jonn You can get or make cords with RJ-45 in one end and RJ-11 in the other. http://www.connectworld.net/cgi-bin/dataw/L0531 7 *$5.16 Leif * ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Get back in dialplan with number-parsing
I'd like to put a phone in a special context, where a test is made on its
business hours, then if so, proceed to the normal context to do whatever it
does with outgoing and local calls.
I've tried, just to go from one context to the next:
[specialoutgoing]
exten = _X.,1,noop(This is a special content)
exten = _X.,n,gotoiftime(?forbidden,1)
exten = _X.,n,goto(outgoing,${CALLERID(dnid)},1)
I use _X. to match anything, but if the call is allowed, I want to jump back in
the [outgoing] context and restart parsing the dialled number.
exten = _X.,n,goto(outgoing,${CALLERID(dnid)},1)
works only id the dialled extension exists precicely in outgoing context, not
in included contexts, and does not to pattern matching.
I can't include [outgoing] in [specialoutgoing], because the number has already
been matched by _X.
I don't want to rewrite the whole dialplan in [specialgoing] or to put the test
into the existing contexts.
Leif
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Re: [asterisk-users] Slightly OT - Oreka Call Recording
- Original Message - From: Tim Nelson To: asterisk-users@lists.digium.com Sent: Wednesday, December 02, 2009 12:06 AM Subject: [asterisk-users] Slightly OT - Oreka Call Recording Greetings all- I'd like to install Oreka on a Centos 5.x server for monitoring my Asterisk systems(using port mirroring) but I find I'm having problems with the version of libpcap installed. /me wonders why you don't let asterisk record the audio itself instead of adding a 3.rd party 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
[asterisk-users] Patch for app_dial.c: exit when just one ext is busy.
I made a patch to app_dial.c to make Dial(ext1ext2ext3,tumeout,B)
return busy when just one extension is busy.
http://www.neland.dk/app_dial.c.diff
It works, but...
I can't figure out setting/reading an option.
It looks fairly easy, but the flag is always set.
*** app_dial.c.org 2009-11-04 22:15:50.0 +0100
--- app_dial.c 2009-12-01 09:29:19.0 +0100
***
*** 98,103
--- 98,105
however, the variable will be unset after use.\n\n
Options:\n
A(x) - Play an announcement to the called party, using 'x' as the
file.\n
+ B- When dialling multiple extensions, return BUSY as soon as
one \n
+extension is BUSY.\n
C- Reset the CDR for this call.\n
c- If DIAL cancels this call, always set the flag to tell the
channel\n
driver that the call is answered elsewhere.\n
***
*** 283,288
--- 285,291
#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)
enum {
OPT_ARG_ANNOUNCE = 0,
***
*** 302,307
--- 305,311
AST_APP_OPTIONS(dial_exec_options, BEGIN_OPTIONS
AST_APP_OPTION_ARG('A', OPT_ANNOUNCE, OPT_ARG_ANNOUNCE),
+ AST_APP_OPTION('B', OPT_SINGLE_BUSY),
AST_APP_OPTION('C', OPT_RESETCDR),
AST_APP_OPTION('c', OPT_CANCEL_ELSEWHERE),
AST_APP_OPTION('d', OPT_DTMF_EXIT),
***
*** 626,635
--- 630,650
watchers[pos++] = in;
for (o = outgoing; o; o = o-next) {
/* Keep track of important channels */
+ if (ast_test_flag64(o, OPT_SINGLE_BUSY))
+ ast_verb(2, OPT_SINGLE_BUSY set\n); /*
always set, why? */
if (ast_test_flag64(o, DIAL_STILLGOING) o-chan)
watchers[pos++] = o-chan;
numlines++;
}
+ /* I'd like to test for OPT_SINGLE_BUSY set, but I can't
figure it out /*
+ /* if (ast_test_flag64(outgoing,OPT_SINGLE_BUSY)
num.busy) doesn't work, the flag is always set */
+ if (1 num.busy) {
+ ast_verb(2, One channel was busy, won't try the
others\n);
+ strcpy(pa-status, BUSY);
+ *to = 0;
+ return NULL;
+ }
+
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);
Anybody wanna look into it?
Leif
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Re: [asterisk-users] Please some enlightment on ENUM !!
Norbert Zawodsky skrev: 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. You probably have too many no/nots :-) 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) Invent all you want, nobody can call those fantasy-numbers anyway. Perhaps, a fraction of a percent, who are using ENUM. Perhaps your voisp directs extra digits to you, but pstn-exchanges have a dialplan, starting to dial when the standard number of digits is entered. 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 Don't rely on it. :-) ___ -- 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] Patch for app_dial.c: exit when just one ext is busy.
Rob Hillis wrote: Leif Neland wrote: I made a patch to app_dial.c to make Dial(ext1ext2ext3,tumeout,B) return busy when just one extension is busy. Forgive me for the question, but /why/ do you want this behaviour? Isn't the whole point of dialling multiple extensions so that a call has a greater chance of being answered? Because I might have more phones than mouths :-) If I'm busy with one conversation, I don't want to hear another phone ring. I might have a desktop and a portable phone. There were a similar wish for queues a month ago or so: Given an office/warehouse/home with several phones, but fewer persons. Queue should only have so many active calls at the same time; eg two persons in the office, 5 phones. two calls come in, ringing every phone, two calls gets picked up. Third incomming call stays in queue until one conversation is terminated. 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] Patch for app_dial.c: exit when just one ext is busy.
Magnus Benngård wrote: Hi! Would be a very nice feature for example the following scenario: Me has 2 phones, one ordinary SIP phone attached to the SIP server and one Cell phone. If someone calls my extension it will ring in both, but if I talk in for example the SIP phone I dont want it to ring on my cell phone. On Tue, 01 Dec 2009 21:35:11 +1100, Rob Hillis r...@hillis.dyndns.org wrote: Leif Neland wrote: I made a patch to app_dial.c to make Dial(ext1ext2ext3,tumeout,B) return busy when just one extension is busy. Forgive me for the question, but /why/ do you want this behaviour? Isn't the whole point of dialling multiple extensions so that a call has a greater chance of being answered? ___ I sure hope it will be implementet. Just need somebody a little more fluent in C than me, to fix the parameter-stuff :-) 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 ___ -- 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] Please some enlightment on ENUM !!
Philipp Kempgen wrote: Leif Neland schrieb: Norbert Zawodsky skrev: 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) Invent all you want, nobody can call those fantasy-numbers anyway. Perhaps, a fraction of a percent, who are using ENUM. Leif, ever heard of direct inward dialing and PRI? http://en.wikipedia.org/wiki/Direct_inward_dialing http://en.wikipedia.org/wiki/Primary_rate_interface You can actually own a block of numbers like 01234567. You are free to map these DID numbers to extensions or do what ever you like. And it is guaranteed that nothing in the 01234567... range will ever be assigned to a different PSTN subscriber. Ok ok, I may have been too harsh... Here in Denmark, when you have the number 12345678, that's it, you don't get 12345678xxx That it's different in OP's country Austria, I didn't know. In that case, it makes perfectly sense to subdelegate enum, although perhaps it should just be a wildcard, so every number in the 12345678-domain goes to the same entrypoint just as a pstn-call does. Otherwise it could be confusing that when extension 10 changes physical sip-adress, you also have to remember to change the enum. Even more, enum-ing a number directly to a sip-phone sounds impractical. I know about blocks. When 3 major banks in Denmark joined in the end of last century, they got the block with being their main number, and the 'es for extensions. 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] Max how many users in sip.conf
mtha...@gmail.com wrote: later i figured out the following. my sip.conf was 2.2Mega Bytes size when populated with 50k users. That means 2.2 x 1024 = 2.2 GB of memory. which is definitely not an option with my small amazon system. I tried with 20k users, hola.. everything works fine. tried with 25k, still works, but asterisk took time to load sip module. tried with 40k, asterisk couldn't start. so decided to settle with 25k userbase. Perhaps you should look into realtime; storing the users in MySql database instead. 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] Patch for app_dial.c: exit when just one ext is busy.
Leif Madsen wrote: Leif Neland wrote: I made a patch to app_dial.c to make Dial(ext1ext2ext3,tumeout,B) return busy when just one extension is busy. In order to have your patch considered at all, you will need to file an issue in the issue tracker and attach your file to it after signing the license agreement. Otherwise, the developers (at least at Digium) won't look at the code or be able to offer any feedback. I didn't think the patch was ready to submit to the issue tracker, before it was working... And right now, I don't understand how to read an option flag, it seems. 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] 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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[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] 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
[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] ((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] Max how many users in sip.conf
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 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] Please some enlightment on ENUM !!
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. 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? But if a pstn or cell call +431123456720 will it be connected to +4311234567 ? Or will the call fail? If so, +431123456720 is an invalid number. 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] Please some enlightment on ENUM !!
- Original Message - From: Norbert Zawodsky To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Thursday, November 26, 2009 10:46 AM Subject: Re: [asterisk-users] Please some enlightment on ENUM !! Leif Neland schrieb: But if a pstn or cell call +431123456720 will it be connected to +4311234567 ? Or will the call fail? If so, +431123456720 is an invalid number. Leif That depends on the Dialplan coding. A non-sip call comes in from the VoIP provider into the associated context. The provider ignores anyway if there is an extension specified, or not. He just routes any call to my base number to me and the dialplan decides how to handle nonexistent extensions... Ymmw. If I make a cell call with extra digits beyound the 8 digts to my asterisk, the extra digits are stripped off, I only see the official 8 digits in the sip dialog, not the extension. To call in directly, I use 12345678p911 in my cell. 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] Please some enlightment on ENUM !!
- Original Message - From: Norbert Zawodsky To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Thursday, November 26, 2009 10:46 AM Subject: Re: [asterisk-users] Please some enlightment on ENUM !! Leif Neland schrieb: But if a pstn or cell call +431123456720 will it be connected to +4311234567 ? Or will the call fail? If so, +431123456720 is an invalid number. Leif That depends on the Dialplan coding. A non-sip call comes in from the VoIP provider into the associated context. The provider ignores anyway if there is an extension specified, or not. He just routes any call to my base number to me and the dialplan decides how to handle nonexistent extensions... Ymmw. If I make a cell call with extra digits beyound the 8 digts to my asterisk, the extra digits are stripped off, I only see the official 8 digits in the sip dialog, not the extension. To call in directly, I use 12345678p911 in my cell. 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] Please some enlightment on ENUM !!
Norbert Zawodsky skrev: SIP schrieb: Yes... you would have to register (and possibly pay for, dependent on the ENUM registrar) each individual number. The idea behind ENUM is that it's an E164 number that is already yours that maps to whatever you want it to map to (email, SIP, etc). The key point here is that you already own the E164 number. If you do, then you could register them all at e164.org for free. If you don't own the individual numbers, you shouldn't be allowed to register them as your own. That sort of breaks the ENUM concept of a number you take with you as a personal identifier. 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 Hi N. ! Thanks for your answer. Either I don't understand what you want to tell me or this thread slowly drifts away from my original question. My original question was: If you own a telephone number which connects to your company and you have a PBX (like asterisk) and some extesniosn behind that, how/where do you enum-register each extension so that each extension can be reached from outside by a SIP uri? Meanwhile I managed to speak to a technician at my-enum.at, which is my registrar at e164.arpa. He *comfirmed* my original assumption: If you have a telephone number and want to paticipate in enum, you have to register that number at - for example - e164.arpa. If you operate extensions behind that number and you want them to be reachable too, you have to run your own DNS server and register this server at e164.arpa. This server is naturally under your responsibility and you manage all your extension yourself. It is works exactly like any other DNS resolution. 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
Re: [asterisk-users] Please some enlightment on ENUM !!
- Original Message - From: Norbert Zawodsky To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Monday, November 23, 2009 3:15 PM Subject: [asterisk-users] Please some enlightment on ENUM !! Hello all you Gurus out there! Please could you explain something to me: Currently I try to get ENUMLOOKUP() working. Naturally I do all the testing with my own number. I registered my number at e164.org I paid for registration of my number at a registration agent for e164.arpa (I know, I don't need both. I just did the .arpa registration first and later discoverd the free .org service) Assume my number was +4311234567 dig 7.6.5.4.3.2.1.1.3.4.e164.org and dig 7.6.5.4.3.2.1.1.3.4.e164.arpa both return NAPTR records. Now for the less clearer points: Your'e supposed to register your number without any extension. If I have some extensions here, how can the calling party get the correct sip uri to the requested extension? Do I have to run my own DNS server in that case? If for example if someone wants to call extension 10, is the ENUMLOOKUP(431123456710) request forwarded to my local DNS server by the e164.arpa server? Or how does that work? If everybody supported enum, it might be usefull to register extension 10 in enum, otherwise: Your extension 10 must have its own phonenumber, to be able to dial it directly. Just as with ordinary pabx. Eg: 123 555 is the reception 123 555 0010 is extension 10 Just some ideas: Is there free (as in not connected to a voisp) numbers, which can be registered in enum? Then you could use those numbers for extensions. But they would only be callable by enum. If the calling of extensions is only to be used by knowledgeable friends you could have them add your own enum-domain to their setup. 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] Please some enlightment on ENUM !!
Norbert Zawodsky wrote: Leif Neland schrieb: - Original Message - *From:* Norbert Zawodsky mailto:norb...@zawodsky.at *To:* Asterisk Users Mailing List - Non-Commercial Discussion mailto:asterisk-users@lists.digium.com *Sent:* Monday, November 23, 2009 3:15 PM *Subject:* [asterisk-users] Please some enlightment on ENUM !! Hello all you Gurus out there! Please could you explain something to me: Currently I try to get ENUMLOOKUP() working. Naturally I do all the testing with my own number. I registered my number at e164.org I paid for registration of my number at a registration agent for e164.arpa (I know, I don't need both. I just did the .arpa registration first and later discoverd the free .org service) Assume my number was +4311234567 dig 7.6.5.4.3.2.1.1.3.4.e164.org and dig 7.6.5.4.3.2.1.1.3.4.e164.arpa both return NAPTR records. Now for the less clearer points: Your'e supposed to register your number without any extension. If I have some extensions here, how can the calling party get the correct sip uri to the requested extension? Do I have to run my own DNS server in that case? If for example if someone wants to call extension 10, is the ENUMLOOKUP(431123456710) request forwarded to my local DNS server by the e164.arpa server? Or how does that work? If everybody supported enum, it might be usefull to register extension 10 in enum, otherwise: Your extension 10 must have its own phonenumber, to be able to dial it directly. Just as with ordinary pabx. Eg: 123 555 is the reception 123 555 0010 is extension 10 Just some ideas: Is there free (as in not connected to a voisp) numbers, which can be registered in enum? Then you could use those numbers for extensions. But they would only be callable by enum. If the calling of extensions is only to be used by knowledgeable friends you could have them add your own enum-domain to their setup. Leif Hi Leif! No, I cannot believe that this was the right way. It would mean that I would have to register ( pay !!) for every single extension. Just as you would have to pay for a bunch of PSTN-numbers if people should be able to call into the extensions via PSTN As ENUM is implemented, it is a mapping of PSTN-numbers to routing. There is not an option to further delegate numbers below a PSTN-number to extensions. But people can dial into your Asterisk via ENUM, and then dial the extension at the voice prompt. However, at e164.org, you can get a FREE164 number out of the +882 99 number pool. These numbers are for dialing Internet hosts only, they are not 'real' telephone numbers! There you can get a series like 88299 008971 0 to 88299 008971 for your extensions. 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] Prevent Dial if any extension is busy
Magnus Benngård skrev:
Hi!
Part of extensions.conf:
exten = 985,1,Dial(SIP/0317998985H323/00702221...@avaya,20)
exten = 985,2,Goto(985-${DIALSTATUS},1)
exten = 985-BUSY,1,VoiceMail(0317998...@inputinterior.se,b)
exten = 985-BUSY,2,PlayBack(vm-goodbye)
exten = 985-BUSY,3,HangUp()
exten = 985-NOANSWER,1,VoiceMail(0317998...@inputinterior.se,u)
exten = 985-NOANSWER,2,PlayBack(vm-goodbye)
exten = 985-NOANSWER,3,HangUp()
0317998985 is a direct connected SIP phone
0702221448 is a celluar phone.
When dialing 985 both phones rings, perfect
If none answer within 20 seconds,
VoiceMail(0317998...@inputinterior.se,u), perfect
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.
If the cell was only called via asterisk, you could set a flag, when
asterisk called extension 985, and clear it, when hanging up, but I
guess the phone is used for call out via regular cell service, and also
called directly on its own number.
You don't own the cell-company, and can setup an API to get the status
of the cell, right? I didn't think so :-)
You could do this:
check if sip is busy, using ChanIsAvail
If so, go to voicemail.
Else, dial cell, timeout 20 sec
if busy go to voicemail
else dial sip, timeout 20 sec
if not answered. go to voicemail.
But this will give 20 seconds delay before sip rings, and 40 seconds
timeout for the caller before voicemail.
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
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Re: [asterisk-users] softphone/debug panel with BLF
Philipp Kempgen skrev: Leif Neland schrieb: Philipp Kempgen skrev: Leif Neland schrieb: Mostly to debug/test BLF, is there a softphone or another app. which can subscribe to hints on Asterisk? X-Lite It does not subscribe to hints on Asterisk. It does. In the contact drawer: Add contact - Contact Methods: Softphone, Phone/Address = Extension, tick Show this contact's availability. I stand corrected. Indeed it does. Now I just have to discover why I can't subscribe to the asterisk at work from home. 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] softphone/debug panel with BLF
Philipp Kempgen skrev: Leif Neland schrieb: Mostly to debug/test BLF, is there a softphone or another app. which can subscribe to hints on Asterisk? X-Lite? http://www.counterpath.com/x-lite.html Philipp Kempgen It does not subscribe to hints on Asterisk. 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] clever ways to share an extension between sip and fxs
Ira skrev: At 07:06 AM 11/18/2009, you wrote: I know that I'm not looking for Dial(SIP/xSIP/y) - as documented, this handles nothing like what I'm looking for. It's not the answer you're looking for, but that feature is built into a Aastra 480i-CT and I think a 57i-CT. Do you know if this phone can also connect to a dect headset ? (which currently connects to a Siemens base, which also supports standard dect phones) ___ -- 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] BLF with SPA941?
Leif Neland wrote: - Original Message - *From:* Ex Vito mailto:ex.vitor...@gmail.com *To:* Asterisk Users Mailing List - Non-Commercial Discussion mailto:asterisk-users@lists.digium.com *Sent:* Thursday, November 12, 2009 3:59 PM *Subject:* Re: [asterisk-users] BLF with SPA941? Although I've never tested such feature on those devices, I know that it was only enabled in a recent firmware (6.1.3a/6.1.5a ?). Are you running it ? Appearently, the latest firmware for SPA9x1 is 5.1.8, the SPA9x1 is not receiving the 5.2. and 6.1 firmware. There is a somewhat heated discussion here, which I unfortunately didn't read before ordering. https://www.myciscocommunity.com/thread/1541 SPA9X1 are our entry-level business IP Phones. Their feature set will not evolve (no new features are expected on this series) from what's existing today (5.1.8). For LDAP and other features we recommend the SPA9X2 product family, which is our mainstream SIP Small Business IP phones family. But I'm not ready to try forcing a SPA9x2-software into a SPA941. I tried downloading SPA942 firmware to SPA941, it didn't accept it. 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
[asterisk-users] *1.4 Received SIP subscribe for unknown event package: call-info
I've got a SPA942 subscribing to hints to a local asterisk 1.6; this works. But when I try to subscribe to a remote asterisk 1.4, it doesn't work; the BLF is flashing yellow. I see this in the log: Received SIP subscribe for unknown event package: call-info The SPA942 extended function for the key is fnc=blf+sd+cp;sub...@hostname.of.remote Is Asterisk 1.6 different in 1.4 regarding BLF? 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
[asterisk-users] softphone/debug panel with BLF
Mostly to debug/test BLF, is there a softphone or another app. which can subscribe to hints on Asterisk? Heck, it doesn't even need to be able to do calls :-) 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] BLF with SPA941?
- Original Message - From: Ex Vito To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Thursday, November 12, 2009 3:59 PM Subject: Re: [asterisk-users] BLF with SPA941? Although I've never tested such feature on those devices, I know that it was only enabled in a recent firmware (6.1.3a/6.1.5a ?). Are you running it ? Appearently, the latest firmware for SPA9x1 is 5.1.8, the SPA9x1 is not receiving the 5.2. and 6.1 firmware. There is a somewhat heated discussion here, which I unfortunately didn't read before ordering. https://www.myciscocommunity.com/thread/1541 SPA9X1 are our entry-level business IP Phones. Their feature set will not evolve (no new features are expected on this series) from what's existing today (5.1.8). For LDAP and other features we recommend the SPA9X2 product family, which is our mainstream SIP Small Business IP phones family. I am still thinking this i a political decision, and the SPA9x1 would be able to support the BLF etc features. Reminds me of a rumour that an upgrade to some photocopiers to offer more features, costing several k$, were simply a technician cutting a jumper; the hardware were already ready. But I'm not ready to try forcing a SPA9x2-software into a SPA941. 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] allowguest defaults to yes for SIP
I think just renaming the [default] to [public] or [unautorized], and a comment saying Don't put outgoing calls in this context, as unauthorized users, even from outside, are routed here by default. would be enough. I'm not sure if local phones should automatically be routed to a [local] context. I think the [public] should be available for guest users, and be published, or at least be in the enum database. Why should my call (and my money) go from my desk via my ip-pabc to my voisp possibly through pstn (through echelon) to your voisp to your ip-pabc to your desk, when it could go from my ip-pabc to your ip-pabc directly. 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] CALLER-ID, MUSIC ON HOLD, HOLD, QUEUE
- Original Message - From: aster...@opensourcesolution.in To: asterisk-users@lists.digium.com Sent: Friday, November 13, 2009 9:47 AM Subject: [asterisk-users] CALLER-ID, MUSIC ON HOLD, HOLD, QUEUE hi all, i had installed and configured asterisk on centos 5.3, i had made a minimum dial plan in which i had made two extentions. i am easily able to make call from one extention to other extention. i know its just a basic thing which i had done n i had done from this place only. now i want to features of dial plan.i want to implement these features in my dial plan. HOLD MUSIC ON HOLD CALLER-ID QUEUE guys ur help n support will be highly appreciated. There are many fine explanations on the net. Read and try, if you then have problems with the details, come back. Or you can pay a consultant to do your work 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
[asterisk-users] BLF with SPA941?
Appearently SPA941 is less than a SPA942 without two ports, poe and backlight. There is less features too, it doesn't support BLF. Is it possible to hack 942-software into 941, or is there another workaround? 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] Incoming Call Ring
- Original Message - From: Dan Journo To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Thursday, November 12, 2009 1:24 PM Subject: [asterisk-users] Incoming Call Ring Hello, I have Asterisk set up with 6 extensions. When a call comes in, I use a Dial command to call all the extensions together until someone picks up. The problem is, when there is an incoming call and an extension is in use, if the extension puts down the phone while the incoming call is still ringing, that extension doesn't ring. This is because when the Dial command was executed, that extension was busy. Is there any way to make that extension ring as soon as its available if there is still an incoming call? You could put all 6 phones in a queue, and call that instead. But there will still be a delay before Asterisk calls the phone again. You could put the phones in a pickup-group, and the user could pick up the call, default is *8 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] Networking Concept
- Original Message - From: B.Masoud @ SH To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Sent: Tuesday, October 06, 2009 1:14 AM Subject: [asterisk-users] Networking Concept Hello, I would like to know how Asterisk deal in this case: Assume I have a Main Asterisk Server located in UK, and another box that have PSTN interfaces located in China, now the purpose is to FW calls through PSTN. Assuming I have a client who is calling from Japan to my main switch in UK and he is calling China, (japan have latency around 500ms to UK and 100ms to China), how asterisk will deal with this call?? Will his latency be JAPN-UK + UK-China (around 1000ms !) or only from Japan to China??? Be sure not to run into trouble for running inlicenced ip-telephony in China, so the government can't (as easily) intercept your calls. Leif ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Who am xxx talking to.agi
In relation to our CRM-system I'd like to send a query to asterisk who is extension xxx talking to. When the operator enters the page with customer data, the crm should send a query to asterisk, to get the cli of the call the operator is having. If the number is matching the customers number in crm, a record will be made, if it is not, a popup Are you talking with this customer now?, if confirmed, the number will be recorded in the crm. Can asterisk answer this question? I've tried using sip show channels and sip show peer, but the cli is not in an obvious place. Is it better done by parsing logfile or storing numbers in the internal database from the dialplan? Leif ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Wrong hint, ringing when idle. after hangup.
I have 3 phones, SIP/3, SIP/6 and SIP/9 SIP/3 subscribes on hint on SIP/9 Phone 6 calls phone 9, blf on phone 3 flashes until 9 picks up, then it is steady red. That's correct. But when 9 hangs up the hint goes to InUseRinging, the light on 3 is still flashing. It keeps flashing until somebody calls 9 and hangs up again. -- Executing [...@local:1] Dial(SIP/6-35236014, SIP/9,120) in new stack -- Called 9 == Extension Changed 9[hintcontext] new state Ringing for Notify User 3 -- SIP/9-36a07014 is ringing == Extension Changed 9[hintcontext] new state InUse for Notify User 3 -- SIP/9-36a07014 answered SIP/6-35236014 -- Packet2Packet bridging SIP/6-35236014 and SIP/9-36a07014 -- Executing [...@local:1] Hangup(SIP/6-35236014, ) in new stack == Spawn extension (local, h, 1) exited non-zero on 'SIP/6-35236014' == Spawn extension (local, 9, 1) exited non-zero on 'SIP/6-35236014' -- Executing [...@local:1] Hangup(SIP/6-35236014, ) in new stack == Spawn extension (local, h, 1) exited non-zero on 'SIP/6-35236014' == Extension Changed 9[hintcontext] new state Idle for Notify User 3 == Extension Changed 9[hintcontext] new state InUseRinging for Notify User 3 (queued) == Extension Changed 9[hintcontext] new state InUseRinging for Notify User 3 Asterisk 1.6.0.15 built by root @ arnold.neland.dk on a i386 running FreeBSD on 2009-09-29 07:49:45 UTC Leif ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] callfile to auto-answering extension
I have a SPA742, which can autoanswer a call In the dialplan, I have this: exten = 28,1,SIPAddHeader(Call-Info:\;answer-after=0) exten = 28,2,dial(SIP/36) Now I want some external event initiate a call to that phone and play a message. I have been thinking of dialfiles, but I believe there is a problem: Dialfiles call a channel, and then executes the dialplan. I need to SIPAddHeader first, then make the call. Or am I missing something obvious? Can I, via a callfile, or command-line parameters to Asterisk start a dialplan-script? eg asterisk -someflag execute callalert then in dialplan [callalert] exten = s,1,SIPAddHeader(Call-Info:\;answer-after=0) exten = s,2,dial(SIP/36) exten = s,3,Playback(firealert) exten = s,4,Hangup Leif ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] callfile to auto-answering extension
Ex Vito skrev: 2009/9/27 Leif Neland le...@neland.dk: Can I, via a callfile, or command-line parameters to Asterisk start a dialplan-script? eg asterisk -someflag execute callalert then in dialplan [callalert] exten = s,1,SIPAddHeader(Call-Info:\;answer-after=0) exten = s,2,dial(SIP/36) exten = s,3,Playback(firealert) exten = s,4,Hangup ...sure, use Local channels. You can use Local/ext@context as the originating channel in a call file or AMI/CLI originate command. -- Sorry, I'm a little rusty... What exactly do I write, If I want to use a CLI originate command, to execute the above callalert? Leif ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Suppress MusicOnHold in Queue
(catching up while my adsl is offline) David L. West wrote: I want callers to go into the queue(s) and just hear ringing instead of MOH. Is this possible? If everything else fails, you can generate a file with ringing tones, and use that for moh. 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] Single ringer phone for incoming calls, that anyone can answer
(While my adsl is down, I'm reading old posts.) Tom Lanyon wrote: Hi list, Does anyone have any advice on the following: Incoming calls to our office come in on a SIP trunk. Since all our offices/desks are in close proximity, we would like just a single phone to ring when a call comes in instead of ringing every person's phone. Currently we've got this working by having all the phones in a callgroup/pickupgroup and incoming calls ring the 'ringer phone' extension, then we can use the *8 to pickup the incoming call from any other phone. The problem though, is that if two people in the office call each other, *8 from a third phone also picks up their call, which is not the desired effect. Use the application Pickup exten = 88,1,Pickup(SIP/singleringerphone) -= Info about application 'Pickup' =- [Synopsis] Directed Call Pickup [Description] Pickup([EMAIL PROTECTED][EMAIL PROTECTED]): This application can pickup any ringing channel that is calling the specified extension. If no context is specified, the current context will be used. If you use the special string PICKUPMARK for the context parameter, for example [EMAIL PROTECTED], this application tries to find a channel which has defined a channel variable with the same content as extension. Leif So in essence, I'm asking whether there's a better way to pickup an incoming call from our external SIP trunk, whilst its ringing only a specific extension, without picking up overlapping internal calls? Regards, Tom ___ --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] Dell poweredge 860 acceptable for officeenvironment ?
Steve Totaro wrote: Stephen Bosch wrote: Olivier wrote: I'm really after 1U-2U silent servers as I've got the feeling most of them are too noisy for offices and most of our clients don't have server rooms. Try this: http://www.tomshardware.com/2006/01/09/strip_out_the_fans/ -s The fans are in there for a reason. It appears you haven't read the article. The tomshardware-guys (no gals would do this...) have removed the fans, and immersed the innards of the computer in a sealed cabinet filled with cooking oil. So they have a completely silent machine in 40C warm oil. Amazing... Leif ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] camp on off-line phone
When phone A registers, I want phone B to ring, when picked up, it should call phone A and connect the phones. Translated: When GF in Mexico powers up laptop where soft iax-phone registers automatically, I want to talk to her asap :-) How to? Leif ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ringing oddity/stupidity
J. Oquendo wrote: Anyone experience ring oddities with extensions.conf rollovers? Let me summarize... One of my extensions.conf file is built to ring during the day, ring/go to voicemail after a certain time: [main-aa] exten = s,1,GotoIfTime(17:00-8:30|mon-fri|*|*|*?main-night-aa,s,1) exten = s,2,GotoIfTime(*|sat-sun|*|*|*?main-night-aa,s,1) ... [main-night-aa] exten = s,1,Answer exten = s,2,Background(/etc/asterisk/night) exten = s,3,Voicemail([EMAIL PROTECTED]) exten = s,4,Hangup When in night mode, if someone called, while Asterisk would show the phone as ringing (and INDEED the phone would ring) the caller wouldn't hear the phone ring. No music, no ringing no thing until the amount of time the rings ran out and then be transferred into voicemail. So... (un)Leet ASCII explanation: Caller (after hours) -- Dials in -- Press extension -- Asterisk makes transfer -- Caller hears dead air -- No one answers -- Voicemail -- Caller now hears voicemail prompts According to the dialplan, there should be no ring at all, it should go directly to voicemail. How long is the Caller hears dead air -- No one answers time? To comfort the caller you could add exten = s,1,ringing exten = s,2,wait(2) exten = s,3,answer() exten = s,4,Background(/etc/asterisk/night) exten = s,5,Voicemail([EMAIL PROTECTED]) Leif ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ATCOM AT 468 manuals and firmware anyone?
Erick Perez wrote: Hi there, im looking for another place that provides manuals and firmware updates for the ATCOM AT 468 and their configuration with asterisk. the site www.atcom.com.cn has non functional download links. I suppose you mean the AG 468 If you can find somebody who still uses Internet Explorer, the links works. The download page used to have a link for a page which worked in Firefox, but not anymore. But anyway, here are the links. http://atcom.com.cn/en/down/userguide/EN/AG-468/AG468_User_ManualGuide.rar http://atcom.com.cn/en/down/program/en/ng_series/ag468_060119_vr41rls.zip Leif ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Adding 4 more POTS lines
Jim Freeze wrote: Hello I have a working * server with a TDM card and 4 FXO ports. We have 4 lines now and need to add 2 more lines (and possibly two more later). I'm wondering the best upgrade path for this situation. The simplest I can invision is adding another TDM400 card with 4 FXO ports, and use 2 now and the remaining 2 later. Are there success stories with using 2 TDM cards? Any info will be appreciated. I have no experience with the TDM cards, but costwise it is not the best solution, in my opinion. A TDM04B (4FXO) cost around $378 at voiplink.com, while a Grandstream GXW-4104 (also 4FXO) cost $250 at voxilla.com., and the 8FXO version costs $400, almost the same as the 4FXO card. Having the pstn-ip conversion outside the server reduces the load and makes an easier install. I'm using the GXW-4104 , and besides it has trouble detecting danish callerid (a standard not used anywhere else in the world...), i have no complaints against it. Imho, ymmw etc. Leif ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Adding 4 more POTS lines
Jim Freeze wrote: Hi Leif On 1/25/07, Leif Neland [EMAIL PROTECTED] wrote: I have no experience with the TDM cards, but costwise it is not the best solution, in my opinion. A TDM04B (4FXO) cost around $378 at voiplink.com, while a Grandstream GXW-4104 (also 4FXO) cost $250 at voxilla.com., and the 8FXO version costs $400, almost the same as the 4FXO card. I suppose that is my alternative - remove the 4FXO card and add an 8FXO card. But I'm not seeing the prices you list. The Digium TDM2402B is listed at $837.00. Am I missing something? http://www.voiplink.com/Digium_TDM2402B_p/digium-tdm2402b.htm You misunderstand me. A Grandstream GXW-4108 with 8FXO cost $400, almost the same as a TDM404B fully populated 4FXO card. Leif ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] pickup call out of menu
Is it possible to pickup a caller, who is in the menus somewhere, for instance he may be lost in the telemarketer torture script? Just like it is possible to pick up a call on a ringing phone. Leif ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] direct transfer in features
I have some siemens wireless ip-phones. There is no problem entering ** which I have configured in features.conf to be transfer. But then it is difficult to enter the extension, because one have to wait the right amount of time before entering the extension. Because we only have few extensions, is it possible to have each transfer-option as a separate feature in features.conf So can I hardwire **1 to transfer to extension 11, **2 to extension 12 *** to park etc. ? Leif ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Forwarding
Nick Ellson wrote: How might you identify a mobile #? (assuming you refer to cellular phones) Now that phone companies are allowing you to transfer your land line to a mobile, it's no longer practical to use prefix blocking. If a land line is transfered to mobile, does it cost more to call it than a real land line? If it does; I'd require the phone company to give me a warning tone when I call a disguised mobile. If it doesn't, then don't bother to block mobiles. Leif ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 911 Testing
Rich Adamson wrote: Dovid Bender wrote: Good Morning List, When setting up a pbx and you want to test your 911 settings do you call 911 and tell them its a test call or do you relly that you set it up properly and hope for the best when some one call's 911 ? I believe most 911 centers would prefer you call their non-emergency number before testing to let them know what you're about to do. They may suggest a less busy time to do the tests, etc. I know a lot of installers that just dial 911 without any previous contact and I don't recall any of them getting chewed out for doing it. Guess if you keep the conversation short its less likely to be a bother. According to what I've read somewhere, at least our 911 (112) has an answering machine, saying Alarm central, one moment and a few seconds delay, before the call actually is signaled to the dispatcher, to filter out misdials and crank calls. So if you hang up quickly, they'll never know or be bothered. Leif ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Multiple Subscriptions to SIP accounts at SameDomain
Original Message From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Sunday, January 29, 2006 1:29 AM Subject: [Asterisk-Users] Multiple Subscriptions to SIP accounts at SameDomain Sorry not to have observed etiquet and lurked here for a bit before wading in with a question but I have an issue that may well be because I dont know enough about what asterisk is actually doing under the hood to understand why I cant do what I want with asterisk. Im hoping that someone can point me in the right direction :-) This is what I have: Mandrake 2006 running Asterisk 1.2.3 - no additional hardware - everything is going to be running via SIP. To enable inbound and outbound connectivity I have been experimenting with using various accounts provided by Gosspitel, Sipgate, aql and others and have found the most sucessful have been those provided by Gossiptel. Herein lies the problem. I need to register about six incoming lines all provided by Gossiptel - half of them to be active within one context and half within another. I have sucessfully registered all the lines within sip.conf as follows: register = username1:password1:[EMAIL PROTECTED] register = username2:password2:[EMAIL PROTECTED] etc and then I created a peer and a user for the sip.gossiptel.com domain, but I now find that any calls that come in to any of these registered accounts all ring the 's' extension within the default context. Thats fine as far as it goes but I need to be able to handle each SIP account in its own context. As a half way house, in the course of testing this I did play with creating extensions for each sip account and directing them thus: register = username1:password1:[EMAIL PROTECTED]/ext1 register = username2:password2:[EMAIL PROTECTED]/ext2 and this works fine as well - inbound calls end up activating the assigned extensions within extensions.conf but the problem remains that these extensions themselves have to be within a single context (in my case the default context). From sip.conf: ;register = [EMAIL PROTECTED]/1234 ; ;Register 2345 at sip provider. Calls from this provider connect to local ;extension 1234 in extensions.conf default context, unless you define ;[mysipprovider.com] in a section below, and configure a context Wild guess: A kludge is if you run your own dns: *.gossiptel.mydom.dom.INCNAMEsip.gossiptel.com. Then register each user to his own domain: register = username1:password1:[EMAIL PROTECTED] register = username2:password2:[EMAIL PROTECTED] Then define [username1.gossiptel.mydom.dom] context=user1context [username2.gossiptel.mydom.dom] context=user2context Otherwise, you should just create a patch to allow the syntax register = user[:secret[:[EMAIL PROTECTED]:port][/context[/extension]] Shouldn't be so hard to do :-) Leif ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Detecting Long PDD
Original Message From: Jean-Michel Hiver [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, January 15, 2006 1:16 PM Subject: [Asterisk-Users] Detecting Long PDD Hi List, I've had some issues with some VoIP providers where either: 1 - There is massive PDD but finally the call goes through 2 - There is massive PDD but the call gets rejected anyways You might start by defining PDD. Most google hits for PDD is about autism... Leif ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Recommendations on a WiFi phone for *?
Original Message From: Jean-Michel Hiver [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, January 10, 2006 2:17 PM Subject: Re: [Asterisk-Users] Recommendations on a WiFi phone for *? Rupert Gregory a écrit : Once you've finished drooling over the UTStarcom you can start drooling over the Linksys WIP330 http://ces.engadget.com/2006/01/07/linksys-wip330-in-da-house-but-you-cant-have-one/ VERY nice phone in my opinion. I dunno... it looks like a cell phone, except it's not one. It would be nice if it was a dual GSM / wifi phones which transparently switch to VoIP when you have a strong enough signal. http://www.paesys.com/en/GSM_Wi-Fi_phone_for_SIP_voice_and_data_GTEK_PWG500.htm Probably it can't transfer a call from voip to gsm or back Leif ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Email2fax big problemo
Original Message From: Andrew Nowrot [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, January 04, 2006 8:53 PM Subject: [Asterisk-Users] Email2fax big problemo Hi, Few days ago I installed Email2fax application on my Asterisk box. I works but not in 100 %. Sometimes (to be certain quite often) I don't receive the whole fax. My fax machine cuts off transmission in 1/2 or 1/3 of the page. I read about it on a wiki and some user lists and people say that this behaviour could be cause be the Ghostscript and the conversion to the tiff format, but when I sent an email with tiff format as a attachment I got the same result (I receive only half of the page). This makes it clear it is not an asterisk-problem, but a problem conveting the email to tiff. Other said the this is cause by the 2.4 kernel and I should change it to 2.6, but I'm not sure if they are right. (I must say that I can't change my kernel because of some other applications). I also read that this situation could be caused by the spandsp library. Doesn't sound so, as the tiff appearently is broken long before asterisk and spandsp sees it. Leif ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] connect more the one phone to ONE sip Acoount
Original Message From: Andreas Koch [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, January 02, 2006 1:03 PM Subject: [Asterisk-Users] connect more the one phone to ONE sip Acoount Hello, how is it possible to connect (register) more the one Phone to One Sip-Acoount. With, for example sipgate.de this is not a special feature, it is common. We have users, what like to have more then one Phone, - Homeoffice and Desk Rigth now if a other phone registers whith the data, the first ist removed. You must use a proxy, for instance SER. According to the wiki, FWD and SIPGATE both run SER coupled with Asterisk SER handles multiple registrations, Asterisk doesn't. Leif ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Semi-OT: porting numbers away
Original Message From: Ross C [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Friday, December 30, 2005 7:18 AM Subject: RE: [Asterisk-Users] Semi-OT: porting numbers away Thanks, but I'm looking for information on porting numbers when the current provider holding the numbers goes out of business and is unreachable. Can I get the numbers? The business has had the same phone number for almost 30 years and definitely can't lose the number due to some provider's instability. I'm sure if a provider was going down, the first thing it would do was selling its cistomers to another company. The vultures would not miss such an oppertunity to get new clients :-) I don't think you need worry. Leif ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Semi-OT: porting numbers away
An interesting wrinkle I'm running against is that you cannot port numbers from a cellular carrier to a landline. i.e. I can't port my cell # to a DID on my PRI. I am not sure if this is just a line of bullshit fed to me from Bell Mobility (Canadian CDMA carrier) but I've not had the time to really dig in. They claim that between cell carriers numbers are portable but not from cell to landline. Another wrinkle: In Denmark, it is not possible to port between landline and cellular, and I hope it never will. Because the caller pays for the entire call, and the rates are approx 10 times higher to call cell than land; the called does not pay for receiving calls. (Except when roaming to another country, then the caller pays the regular rate to the border, and the called pays the rest) One can tell if the number is cell or land, and act accordingly. Eg it is cheaper to call cell - cell than land - cell. So several danish voip-providers are using cellular gateways (a box with ethernet in and a handfull of sim-cards in), to avoid going via pstn. The cell-providers doesn't like that (less revenue), and claim they have the right to terminate the sim-card account... Leif ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] voicemail/privacy system
Original Message From: Eck [EMAIL PROTECTED] To: [EMAIL PROTECTED]; asterisk-users@lists.digium.com Sent: Saturday, December 31, 2005 8:26 PM Subject: RE: [Asterisk-Users] voicemail/privacy system If you dont want to get too stuck into the guts of Asterisk yet, the [EMAIL PROTECTED] distribution can do all you have requested with a one button install web configuration via AMP. Personally I think its a great place to start with asterisk whatever your requirements as it makes a good base without having to go through the drudgery of installing asterisk the requirements/add-ons piecemeal, espically AMP, as the prereqs are a stress! (mumbles something about a, thankfully forgotten, nightmarish FreeBSD Asterisk/AMP install then fades into background, wimpering) :) I installed asterisk from the FreeBSD ports, but then took the config's from the live asterisk cd. That was fairly painless. Leif ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GXP-2000 fw 1.0.1.13 and NTP
Original Message From: Peter Bowyer [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Saturday, December 31, 2005 11:34 AM Subject: [Asterisk-Users] GXP-2000 fw 1.0.1.13 and NTP Hi all Slightly OT but I know a lot of GS experts hang out here - I just upgraded a GXP-2000 to firmware 1.0.1.13 to try out the BLF functionality with Asterisk (which so far works as expected), but as a side-effect the phone won't sync with an NTP server - I've tried different server names (time.nist.gov and pool.ntp.org) and IPs in the config, but it refuses to update the time on the display. Anyone heard of this? Any workarounds (other than go back to 1.0.1.12) ? (Hmmm.. just regressed to 1.0.1.12 and it's still not working - curiouser and curiouser said Alice...) My GS BT101 have also developed problems with sync'ing to my ntp-server. I can see, using tcpdump, that the phone asks my server and gets an answer, but the display is not updated. It used to work, but now it usually doesn't, but strangely, sometime it does... Leif ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk - Gizmo
Original Message From: [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, December 21, 2005 9:14 AM Subject: Re: [Asterisk-Users] Asterisk - Skype anywhere/anyhow? On Tue, 20 Dec 2005, AR Tarzi wrote: could you please tell how it interfaces with Asterisk? Could I receive calls into Asterisk? send calls out? I've just downloaded it and am searching (unsuccessfully) for these on Gizmo's site/software. Gizmo isn't just a soft phone. Like Skype, its a service. Unlike Skype, though, the service is open to the rest of the SIP world. So - to call your Asterisk system from Gizmo, simply tell Gizmo to dial [EMAIL PROTECTED] To call Gizmo from Asterisk, simply tell it to dial SIP/[EMAIL PROTECTED] It 'sort of works'. I can call from gizmo to my *, but the url for incoming is SIP/[EMAIL PROTECTED] DTMF from gizmo does not work If gizmo is dialing into the queue, gizmo doesn't notice the prompts from * (which I can see in the *log), but keeps playing ringtones. But when the phone is answered, gizmo knows. and the connection is made. (The queue works as expected, when I call from eg my cellphone to *) So an Answer() is needed before queue(). Leif ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Blind transfer question
From: Jan Saell [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, November 30, 2005 9:32 AM Subject: Re: [Asterisk-Users] Blind transfer question I did a quick check on the blindxfer config parameter and i cant find any referense to that in the sourcecode for 1.2! The features are defined in ... tada... res/res_features.c ! :-) I've found features are detected most reliably when the phone sends DTMF as sip-events, not via RTP (RFC2833) or in-audio Leif ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Better transfer
I find the transfer functions a little lacking. Examples: I get a call I do an attended transfer, but the called extension never answers/I get impatient/I discover I have dialed the wrong extension. I can not get the call back. If I hangup, the caller is also hung up. I'd prefer the caller to stay online and be ringing my phone again. If I do an attended transfer, and hangup before the 3. part answers, the caller is disconnected. I'd prefer the transfer to be turned into a blind transfer, the caller coming back to me if the called ext is not answering If I do a blind transfer, and the called ext is not answereing, I'd like the call to come back to me. Can this be done in dialplan, or must it be changed in the source? Leif ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Better transfer
- Original Message - From: Patrick [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, December 01, 2005 4:00 PM Subject: Re: [Asterisk-Users] Better transfer On Thu, 2005-12-01 at 15:50 +0100, Leif Neland wrote: I find the transfer functions a little lacking. Examples: I get a call I do an attended transfer, but the called extension never answers/I get impatient/I discover I have dialed the wrong extension. I can not get the call back. Iirc in 1.2 you can get the call back with #0. see features.conf My features.conf.sample doesn't have #0: [featuremap] ;blindxfer = #1; Blind transfer ;disconnect = *0 ; Disconnect ;automon = *1 ; One Touch Record ;atxfer = *2 ; Attended transfer Neither can I see any hints in res/res_features.c Unless disconnect above really means abort transfer Leif ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Truncated CDR records
Original Message From: Innocent Evil [EMAIL PROTECTED] you can use 'w' option with 'Dial' on 1.2.x I don't think w do anything like 'wait', If I am wrong, correct me someone please According to app_dial.c w- Allow the called party to enable recording of the call by sending\nthe DTMF sequence defined for one-touch recording in features.conf.\n W- Allow the calling party to enable recording of the call by sending\nthe DTMF sequence defined for one-touch recording in features.conf.\n; There is a difference between a w tagged on to the number, and a w as an option. The options come last in the dial command, after a | Leif ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SNOM and 1.0.9
On 08:48, Tue 29 Nov 05, [EMAIL PROTECTED] wrote: From memory (at a previous installation) you will need a newer version of Asterisk than 1.09 for the lights to work. on 1.0.9 the lights work. In this way: person is on the phone: light is on Person is not on the phone: light is off since 1.2 the lights will blink when the phone is running and above states work the same. Running? Is that a 3. state? Leif ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] stop asterisk when Idle
I still continue to reboot my asterisk box everyday. I posted a message on November 22, but it was on another thread and no one answered me, so I try again here, where a lot of people told be I was a bad administrator (Like a Windows administrator and I don'0t want to resolve my problem) Actually I would like to resolve my problem, but I am not able to do this, so I ask help to anybody who can help me, and repost my last of 22/11/2005 In short, my problem is that, after one or two days of running, chan oh323 suddendly disappear from asterisk box, without giving any warning / error In example, you type oh323 show stats at 11 o'clock , and get an answer from asterisk, about usage of oh323 At 12, without doing anything to the box or to the asterisk, you type the same command, and you get a No such command 'oh323' (type 'help' for help) If you type help, no oh323 commands are available. If you quit asterisk, (STOP NOW) and restart asterisk , no oh323 channel command is available if you reboot the machine everything is again fine ! It is so a crazy situation that to reboot appears (to me) the best thing (I am sorry about this) If you really need to have oh323, then you should test say every 5 minutes or so, and then shutdown asterisk and reboot. if asterisk -r -x oh323|grep help then echo oh323 missing|mail administrator asterisk -r -x stop now reboot fi Ii is better to disconnect the existing users if they can not use the box without oh323 Do you have some kind of monitoring running? Like Big Brother or nagios? It might be interesting to see when oh323 dies. Perhaps you could also use mrtg to graph usage levels to see if there is some kind of correlation between usage and oh323 fatality Leif ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sound problem, please help!
Original Message From: Esteban Maestre [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Friday, November 25, 2005 11:22 AM Subject: [Asterisk-Users] sound problem, please help! Hi all! I have a strange problem when using asterisk. I have configured asterisk to receive calls (FX0). In my configuration, I want asterisk to play music while I record the caller's speech. Dialup-karaoke? :-) Leif ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ver1.2 installation problem
From: Bartosz Wegrzyn - asterisk [EMAIL PROTECTED] Hi, After I compile asterisk v.1.2 is tells me that last thing to do is to make install. Unfortunately it goes it to loop after I type make install this is the loop: else \ mv include/asterisk/version.h.tmp include/asterisk/version.h ; \ fi rm -f include/asterisk/version.h.tmp Any ideas why? Not why, but I deleted version.h and possibly .depend (IIRC) Leif ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk.conf question
Adrian A wrote: Does anyone know what exactly the option transmit_silence_during_record in asterisk.conf does? Is this useful for voicemail recording? Could the option be named any more explicitly? It does _exactly_ what it says it does. Some providers terminate the connection if nothing is transmitted for x seconds. If asterisk sends nothing while the caller speaks his message, the provider might terminate the call. So asterisk can transmit silence (which is not nothing) during record. Similarly you might have to say yes dear regularly to avoid having the connection terminated while talking to your SO. :-) Leif ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New asterisk management tool
I need a hint: From pbxmanager/doc/INSTALL 2. Install a database adaptor via rubygems. Postgresql, Mysql, and Sqlite3 are all supported and tested to work. Eh... How to install? Leif ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] /spool/outgoing delays
Original Message From: Chris Cahill [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Thursday, November 17, 2005 1:15 PM Subject: [Asterisk-Users] /spool/outgoing delays Hi, I have a rather interesting problem with my Asterisk setup at the moment, and was wondering if anybody could shed any light on it! The system is initiated by placing a call file into /var/spool/asterisk/outgoing. This file calls asterisk, so it is calling itself. The process then goes on to call a few agi scripts, and ends up creating another file (via agi) in the outgoing directory, this one being the one that calls the outside world. Are you *creating* the file in the /outgoing directory? You should create it somewhere else and move it into /outgoing, to prevent asterisk to find an incomplete file. Leif ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] gpx-2000 early dial support
Original Message From: Louis-David Mitterrand [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Friday, November 18, 2005 10:10 AM Subject: [Asterisk-Users] gpx-2000 early dial support The gxp-2000's lack of a dialplan (or did I miss it?) led me to activate its early dial option to avoid pressing Send after dialing. Thus the dialplan is controlled by asterisk. It creates an extension matching problem: exten = _00[1-9].,1,Macro(dialcapi) If I dial 0012 the extension is matched immediately. Is there a way to ask asterisk to wait a few seconds for more digits? You seem to contradict yourself. You want to call a few seconds after the last digit. Why implement it in asterisk, when the phone is capable of doing that by itself. Let the phone decide when these few seconds has expired. Remove the early dial again, and set the timeout in the phone. My Grandstreams have 4 seconds digit timeout. Leif ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hung Zap channels
Original Message From: John Heng [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Friday, November 18, 2005 1:56 AM Subject: [Asterisk-Users] Hung Zap channels Hi all, Once in a while, I've found that the zap channel will get stuck (or blocked) even after the call has ended. The way I've fix this is to issue a soft hangup command for that zap channel. However, I'm not always aware of this until a user tells (or complains to) me. What I would like to know is if there is a way to reset all the zap channels or re-initialize the drivers without restarting Asterisk. If so, I could set up a cron job to do it once or twice a week, in the middle of the night. Any suggestion guys?? To have a channel blocked for ½-1 week would not be good, I think... Can you determine in a script if a channel is hung? Then do a soft hangup on it. Run this in cron. Or regularly do a soft hangup on any channel which haven't had activity in x minutes. But the best solution is naturally to determine why the channel hangs and fix the problem. Leif Cheers J Heng ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] voicemial maxmsg
Original Message From: Joseph Rothstein [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Tuesday, November 15, 2005 10:42 AM Subject: [Asterisk-Users] voicemial maxmsg Has anyone tested the maxmsg parameter in the voicemail.conf file? I am trying to restrict the number of messages for each mailbox, but I can't seem to get this parameter to have any effect. I also could not find a single reference to this parameter on the wiki. If anyone has gotten this to work, or know of another way to restrict the number of allowable messages I would sincerely appreciate the help. Try putting a silly value like -1, then asterisk should complain: Invalid number of messages per folder maxmsg=%s. Using default value %i\n, value, MAXMSG If it doesn't complain asterisk isn't reading your value The default and max is: #define MAXMSG 100 #define MAXMSGLIMIT Leif ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Multiple Outbound SIP Trunks
Original Message From: [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, November 15, 2005 2:27 PM Subject: Re: [Asterisk-Users] Multiple Outbound SIP Trunks On Tue, 15 Nov 2005, Pikoro wrote: There will be no discrimination or routes based on outbound calling, like a certain trunk for international calls, another for local calls, etc... Only a group of 10 SIP trunks to be rotated for all outbound calls. Can you explain what you mean by a SIP trunk? SIP just has addresses - sometimes slightly hidden away in sip.conf behind a SIP peer. So if you Dial(SIP/remotehost/number), a SIP invite is sent to the host IP address defined in the SIP peer in sip.conf. If you Dial(SIP/[EMAIL PROTECTED]) then the invite is sent to the host hostname. Normally it makes no difference to either side how many other calls may already by in progress between the two sides. Some providers allow only one outgoing call at a time. Leif ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Price info in SIP packet?
Is there some way my uplink can tell my * the price of a call, either per timeunit in the conversation at start of the call, or the total cost at the end of the call? I'd like to pass the bill on to the extensions. Leif ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with Cisco local conference and hangup
Original Message From: C F [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, November 14, 2005 4:50 PM Subject: [Asterisk-Users] Problem with Cisco local conference and hangup Cisco 7960 gets a call from zap/1, hits conf to call out on zap/2, then hits join, after a while cisco hangsup, at which point zap/1 and zap/2 can still talk, shouldn't asterisk hangup on all three? That is the way I would prefer it to work. Like an attended transfer. Leif ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with Cisco local conference and hangup
Cisco 7960 gets a call from zap/1, hits conf to call out on zap/2, then hits join, after a while cisco hangsup, at which point zap/1 and zap/2 can still talk, shouldn't asterisk hangup on all three? That is the way I would prefer it to work. Like an attended transfer. I cannot understand why, why not use attended transfer then? Customer calls salesperson. Salesperson need assistance of tech support. Salesperson explain problem to tech with customer online to A: Supply tech with needed information only, so the customer does not need to repeat a long story B: Educate customer to be brief and to the point C: Make sure salesperson relay correct information Salesperson then hangup and let customer and tech settle the problem, But as we know now, this is configurable, so we can have it as we like it. Leif ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
