Re: [asterisk-users] How to generate 183 Session Progress
Thank you Klaus and Martin for your answers! It's very helpful! -- -- -- Marc LEURENT lf...@leurent.eu Le vendredi, 23 octobre 2009 20.51:54, Martin a écrit : > You can call application Progress() from within dialplan and it will > cause the Asterisk to send a SIP reply 183 > on the call that came to Asterisk. > > Martin > > On Fri, Oct 23, 2009 at 6:36 AM, Marc Leurent wrote: > > Hello everybody, > > I have 2 users connected on the same Asterisk server that are connected > > with 2 different Asterisk servers. > > > > For outgoing calls, one in receiving 183 Session Progress and the other > > not! Do you have any idea why? Thanks. > > > > I have tried to understand by myself and in their INVITE they have almost > > the same Allow and Supported SIP Headers > > > > The one that works: > >Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, > > NOTIFY, INFO Supported: replaces, timer > > > > The one that doen't work: > >Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > >Supported: replaces > > > > -- > > -- -- > > Marc LEURENT > > lf...@leurent.eu > > > > ___ > > -- 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] SIP interconnection problem
Tarek Sawah skrev:
> you need to post you SIP.conf and your Extensions.conf so someone can
> have a look at them and see if there is anything missing
> what are the contexts you are using with your peers?
> what is the dial plan triggered when calling your destination number?
Machine 1 ---
iax.conf: ==
[general]
bandwidth=low
disallow=lpc10 ; Icky sound quality... Mr. Roboto.
jitterbuffer=no
forcejitterbuffer=no
autokill=yes
[2200]
type=friend
host=dynamic
context=users
username=2200
secret=none
auth=md5
sip.conf ===
[general]
port=5060
bindaddr=0.0.0.0
disallow=all
allow=alaw ; Allow codecs in order of preference
allow=ulaw
allow=gsm
allow=g726
dtmfmode=rfc2833
register => machine_1:wabo...@192.168.10.77/machine_2
[machine_2]
allow=alaw,ulaw,gsm,g726
host=dynamic
secret=wabooba
type=friend
context=sip_incoming
username=machine_2
extensions.conf ==
[general]
static=yes
writeprotect=no
clearglobalvars=no
[globals]
; The outgoing sip trunk
SIP_TRUNK=192.168.10.77
OUTGOING_PREFIX=0
[default]
include => sip-incoming
include => test
[test]
; Create an extension, 600, for evaluating echo latency.
;
exten => 600,1,Playback(demo-echotest) ; Let them know what's going on
exten => 600,n,Echo ; Do the echo test
exten => 600,n,Playback(demo-echodone) ; Let them know it's over
exten => 600,n,Goto(s,6)
[users]
include => sip-incoming
include => outgoing
include => test
[sip-incoming]
include => agi-async
include => internal
[agi-async]
exten => _01,1,Agi(agi:async)
[internal]
exten => _2XXX,1,NoOp()
exten => _2XXX,n,Dial(IAX2/${EXTEN})
exten => _2XXX,n,Hangup()
[outgoing-agi-async]
exten => _${OUTGOING_PREFIX}.,1,Dial(SIP/${ext...@${sip_trunk})
exten => _${OUTGOING_PREFIX}.,n,Set(CALLERID(name)=reason-${DIALSTATUS})
exten => _${OUTGOING_PREFIX}.,n,Agi(agi:async)
[outgoing]
exten => _${OUTGOING_PREFIX}.,1,Dial(SIP/${SIP_TRUNK}/${EXTEN:1})
exten => _${OUTGOING_PREFIX}.,n,Hangup()
Machine 2
sip.conf ===
[general]
port=5060
bindaddr=0.0.0.0
disallow=all
allow=alaw ; Allow codecs in order of preference
allow=ulaw
allow=gsm
allow=g726
dtmfmode=rfc2833
register => machine_2:wabo...@192.168.10.11/machine_1
[machine_1]
allow=alaw,ulaw,gsm,g726
host=dynamic
secret=wabooba
type=friend
context=sip_incoming
username=machine_1
extensions.conf ==
[globals]
; The outgoing sip trunk
SIP_TRUNK=192.168.10.11
Rest is exactly the same. I have a zoiper connected to each machine and I'm
trying to make a call from Machine 2 to zoiper
on Machine 1:
-- Registered IAX2 '2200' (AUTHENTICATED) at 192.168.10.113:4569
-- Accepting AUTHENTICATED call from 192.168.10.113:
> requested format = gsm,
> requested prefs = (),
> actual format = gsm,
> host prefs = (),
> priority = mine
-- Executing [02...@users:1] Dial("IAX2/2200-1200",
"SIP/192.168.10.11/2200") in new stack
== Using SIP RTP CoS mark 5
-- Called 192.168.10.11/2200
[Oct 26 09:20:25] NOTICE[20248]: chan_sip.c:15031 handle_response_invite:
Failed to authenticate on INVITE to '"2200"
;tag=as6173091f'
-- SIP/192.168.10.11-090c2ea8 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing [02...@users:2] Hangup("IAX2/2200-1200", "") in new stack
== Spawn extension (users, 02200, 2) exited non-zero on 'IAX2/2200-1200'
-- Hungup 'IAX2/2200-1200'
Besides that "sip show peers" on either machine shows the other one correctly
registered, and "iax2 show peers" shows the connected zoiper on each machine.
Ideas, please ??
TIA
/Rob
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[asterisk-users] Common Community for exchange the routes via Asterisk boxes
Hello All; I beleive as we have DUNDI in Asterisk, there should be a common community (a website and so on) where those who has Asterisk boxes, they can exchange traffic between each other. Is there something like this? I would like to have a service provider in low rates, any advise? Regards Bilal ___ -- 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] Common Community for exchange the routes via Asterisk boxes
This question is more appropriate for the asterisk-biz list. There are such clearinghouses for traffic peering indeed. I suspect didx.net is likely to be brought up as an example. bilal ghayyad wrote: > Hello All; > > I beleive as we have DUNDI in Asterisk, there should be a common community (a > website and so on) where those who has Asterisk boxes, they can exchange > traffic between each other. > > Is there something like this? > > I would like to have a service provider in low rates, any advise? > > Regards > Bilal > > > > > ___ > -- 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 -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ -- 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] SIP interconnection problem
Ooops.. forgot. The versions of * are: Machine 1: 1.6.1.4 Machine 2: 1.6.0.5 /Rob ___ -- 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] No tone, one way communcation.
1. When i connected my analog phone to fxs card, i cannot get dial tone what could be the problem? I am using elastix 1.5.2 based on centos 5.2 Final. 2. On my 2 sip softphones using x-lite linux versions, i get one way audio how do i solve this?. This problem is also present when i use a windows version on one end and linux version on other end. Any help will be highly appreciated. ___ -- 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] Call Record.
Sir, I am using Asterisk 1.4 and i want to record incoming call and i want to give record file name will start with extension no. and add date and time. i know it is possible with monitor but the problem is when i have receive incoming call than i have ring 4 extension simultaneously and i want to generate record file only for the extension who pickup that call. i have use following statement to pickup call when all four extension no body pickup the call than it will transfer to extension no. 16. exten => s,1,Dial(SIP/21&SIP/23&SIP/18&SIP/14,40,Tt) exten => s,2,Dial(SIP/16,20,Tt) Please help how i can do that Rajeev. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No tone, one way communcation.
On Mon, Oct 26, 2009 at 05:02:18AM -0700, PATRICK KANGETHE wrote: > 1. When i connected my analog phone to fxs card, i cannot get dial tone what > could be the problem? What is the output of: lsdahdi dahdi_hardware > > I am using elastix 1.5.2 based on centos 5.2 Final. Consider also asking there. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- 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] IAX jitterbufer oddity
Hi, First a confession - The box in question is a 1.2.35 box, so this may be solved in a newer version as I know the JB code is all hugely changed, but... It may be worth checking into. Scenario: - IAX outbound call from Asterisk, which rings okay. - Remote end sends ANSWER, which we immediately ACK. - The ANSWER control packet gets put into the JB (that's how I read the code) - The remote end is clustered, and we receive a TXREQ within 1ms of our ACK - chan_iax2 starts to process the TXREQ correctly. What I think happens at this point is that the ANSWER control frame now leaves the JB in order, but is not processed because the channel state has moved into the new transferring state, so ANSWER has no meaning, app_dial never forwards the ANSWER control event to the calling channel, and the bridge is never fully completed, so it all eventually times out. Disabling the JB in IAX does resolve the issue, but is not ideal. I have tried to follow the code in the various versions 1.2, 1.4 and 1.6, but it is just too complicated. Does anyone know if this was addressed since 1.2, or can it still happen in 1.4 or 1.6? Thanks, Steve ___ -- 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] hangup from which side
So this *should* work?? [outgoing] - exten => s,1,Dial(DAHDI/1/5551212,20) - exten => s,2,Noop(I hung up) - exten => s,3,Hangup - exten => h,1,Noop(you hung up) - exten => h,2,Hangup [incoming] - exten => s,1,Answer - exten => s,2,Noop(I hung up) - exten => s,3,Hangup - exten => h,1,noop(you hung up) - exten => h,2,hangup -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Martin Sent: Friday, October 23, 2009 1:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] hangup from which side if you are debugging visually then look at SIP BYE message ... who sent it first and on PRI who sent the DISCONNECT message first. if you need to know that in the dialplan ... then if the originating channel hanged up then the dialplan should stop executing and go straight to h,1 even if Dial(,,g) is used also there is a channel variable HANGUPCAUSE and you can check what it does on the next step with Dial(,,g) and on h,1 ... since I don't know :) Martin On Thu, Oct 22, 2009 at 12:12 PM, B.Masoud @ SH wrote: > When Asterisk establish a call through an outbound trunk, Is there any way I > can know who hang up the call first? The caller or the party called? > > > > Thanks. > > ___ > -- 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] Concurrent calls including mysql taking lot of time for execution
Thanks for all the information. Benaiad: I will try adding in the hosts file and try it once again, also one more that I was in regards to the harddrive, so I thought of replacing with a SSD with high read and write speeds just to check whether its going to reduce the dealy... Regards Sandesh On Thu, Oct 22, 2009 at 6:20 PM, Benaiad wrote: > > Abdulmnem Benaiad > Almontaha CTO > Almontaha IT Co. > cell: +218 92 5200025 > fax: +218 21 4835263 > www.almontaha.com.ly > > > On Wed, Oct 21, 2009 at 11:57 PM, das sandesh wrote: > >> Hi Matt, >> >> I already used the tuning-primer.sh script to enhance the values for the >> parameters, but still it was being slow to connect when there are lot of >> calls (calls around 150-200 calls). Also I reduced mysql queries in the code >> as well as many other steps, but only problem coming is with repect to the >> connection from asterisk to mysql (also I am using and >> not the dns name).is it better to use any additional mysql server apart >> from this application server? or adding additional hardware would help (like >> dual quad core)? >> >> Thanks >> Sandesh >> >> >> On Wed, Oct 21, 2009 at 3:57 PM, Matt Riddell wrote: >> >>> On 22/10/09 7:30 AM, das sandesh wrote: >>> > Hi, >>> > >>> > I tried getting our server setup for 400-500 simultaneous calls, calls >>> > were going through properly but at around 200-250 calls, mysql (connect >>> > ...) statement was taking at least 5-10 sec to connect to the database. >>> > I optimized all possible parameters in my.cnf: >>> >>> Without knowing what you're optimising you're unlikely to have much luck >>> just setting values. >>> >>> We have had quite good success with the tunish-primer.sh script: >>> >>> http://www.day32.com/MySQL/ >>> http://www.day32.com/MySQL/tuning-primer.sh >>> >>> We run with MySQL at about 500 queries per second with no problems - we >>> don't however use Asterisk's MySQL libraries. >>> >>> -- >>> Cheers, >>> >>> Matt Riddell >>> Director >>> ___ >>> >>> http://www.venturevoip.com/news.php (Daily Asterisk News) >>> http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) >>> http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) >>> >>> > > Regarding Mysql delay, I've faced this problem before in my local network > and this problem has been solved by adding all of my servers IP's and names > in the "/etc/hosts" file. > even when connecting locally, I think , asterisk will use a real IP instead > of 127.0.0.1 when connecting to mysql and mysql will try to resolve it's > name, and this step will take some time. > > Regards. > > -- > Benaiad > > > > ___ > -- 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] OT - mISDN and B410P questions
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Olivier a écrit : | | | 2009/10/24 Jean-Denis Girard mailto:jd.gir...@sysnux.pf>> | | Olivier a écrit : | | Hello, | | | | I'm evaluating to possibility to use chan_misdn as a short term | | workaround, in case latest Dahdi is not stable enough for what we are | | planning to do (we wish to use Junghanns and Digium BRI hardware with | | Asterisk 1.6) . | | Dahdi has been working fine for me for a few months, using a Junghanns | DuoBRI and asterisk-1.6. I have used bristuff before, it usually worked | fine, but separate patch. I also tried chan_misdn but quickly abandoned. | | |> Do you mean you abondoned chan_misdn (1.1.9-1) with 1.6.x on Junghanns |> hardware or are you thinking about another combination ? | |> Did you change because revision-2822-enabled Dahdi ran successfully |> enough that you didn't need to further try with misdn or because you met |> bloking issues ? Sorry if that my post was not clear. I'm using Dahdi because I'm happy with it. It works and is simple to set up: it does not need any external patch / software. Before BRI support in Dahdi, I used BRIstuff, which worked fine, but didn't follow Asterisk releases. And it was more work to set up. I tried mISDN, but then I depended on specific kernel versions. I also had echo problems. So I quickly abandoned mISDN. The conclusion is that I'm very happy with BRI support in Dahdi. I've been waiting for it, probably for more than 5 years ! Regards, - -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 50 10 40 / GSM: +689 79 75 27 -BEGIN PGP SIGNATURE- iEYEARECAAYFAkrlyAsACgkQuu7Rv+oOo/hvTgCfaPXjHD90VpeNlSMH/GPY9bzk jvMAn2hzE3JWJ/iY+LXf7Zj6hfHPcT2R =caOJ -END PGP SIGNATURE- ___ -- 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] Cancel attended transfer
Hi folks, I have a simple question regarding attended transfers. I have some queues where agents take calls and I have configured attended transfers between queues. That is, the agent dials the attended transfer extension that routes it to the aproppiate transfer queue where the second agent answers and they both talk for a while. Finally the transferrer leaves the call with *, connecting the client, which was in hold, with the second agent. Is there a way for the transferrer to *cancel* the operation and return to the original call? If the call is queued and the agent leaves with *, it would end up in a blind transfer for the second agent who takes the transferred call. Is there a feature that can be configured to do that? I'm using asterisk 1.4.22 on a production server. TIA, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center ___ -- 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] state_interface backport issue
It's my understanding that the backport is available now in 1.4.
However, seem to be having some issues with it. Just wondering if I have
everything setup right.
I'm running 1.4.26.2 realtime.
queue_members:
`uniqueid` int(10) unsigned NOT NULL auto_increment,
`membername` varchar(40) default NULL,
`queue_name` varchar(128) default NULL,
`interface` varchar(128) default NULL,
`penalty` int(11) default NULL,
`paused` int(1) default NULL,
`state_interface` varchar(128) NOT NULL,
Data:
1, Name, QUEUENAME, Local/1...@agents/n, 1, , SIP/100
Local agents are setup setup in an 'agents' context.
[agents]
exten => 1050,1,Set(agentsip=${DB(agent_sip/1050)}
exten => 1050,2,Dial(SIP/${agentsip})
Queue shows the agent as unavailable when the SIP device (SIP/100) is
down. (as I would hope)... but shows the agent as available all the
other times.
As a result my CLi is on fire with 'busy' notices, because it's trying
to ring an agent even when they are on a call. If I remove the
state_interface, it shows them as 'busy' in the queue, and doesn't ring
them.
Let's see, what else did I forget? Other details:
sip.conf: limitonpeers=yes
and call-limit=5 on each SIP device
queue.conf: ringinuse=no
Anything else I should look for?
Thanks!
-Rob
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[asterisk-users] Answer call from another device
Hello! I remember a while back I saw a way to answer a call from a device that is not from the one ringing, but I don't remember what how to do it. Any help would be great! Thanks, Elliot ___ -- 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] Answer call from another device
*8 is the default value in features.conf to pick up a ringing line if you are in that ring group. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Elliot Murdock Sent: Monday, October 26, 2009 2:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Answer call from another device Hello! I remember a while back I saw a way to answer a call from a device that is not from the one ringing, but I don't remember what how to do it. Any help would be great! Thanks, Elliot ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DAHDI not detecting RINGING Status on the Channel
I am using an 8 port tdm card and also I implemented a dialer using a
.call file generator. As you know on the .call you specify the channel to
call and then the contex/extension/priority to let dial plan continue when
the call is bridge.
My actual problem is that when the call process starts, asterisk (DAHDI)
sets the channel as answered when the truth is that on the other side the
channel has not started to ring yet. My felling is that the DAHDI driver
cannot detect /understand the signaling of the analog line. Here is the
evidence and configuration settings. I live in Argentina BTW
Console output.
[Oct 22 23:42:03] -- Attempting call on
Local/84776...@outgoing_campaign/n for 5...@queues:2 (Retry 1)
[Oct 22 23:42:03] -- Executing [84776...@outgoing_campaign:1]
Macro("Local/84776...@outgoing_campaign-d2c3,2",
"recordcall|501|84776114")in new stack
[Oct 22 23:42:03] -- Executing [...@macro-recordcall:1]
GotoIf("Local/84776...@outgoing_campaign-d2c3,2", "1?5:2") in new stack
[Oct 22 23:42:03] -- Goto (macro-recordcall,s,5)
[Oct 22 23:42:03] -- Executing [...@macro-recordcall:5]
Set("Local/84776...@outgoing_campaign-d2c3,2",
"FILEREC=2009-10-22-23-42-03-SRC-501-DST-84776114") in new stack
[Oct 22 23:42:03] -- Executing [...@macro-recordcall:6]
Set("Local/84776...@outgoing_campaign-d2c3,2",
"FILE_PATH=2009/10/22/2009-10-22-23-42-03-SRC-501-DST-84776114") in new
stack
[Oct 22 23:42:03] -- Executing [...@macro-recordcall:7]
Set("Local/84776...@outgoing_campaign-d2c3,2",
"CDR(userfield)=2009-10-22-23-42-03-SRC-501-DST-84776114.wav") in new
stack
[Oct 22 23:42:03] -- Executing [...@macro-recordcall:8]
MixMonitor("Local/84776...@outgoing_campaign-d2c3,2",
"/opt/rec/2009/10/22/2009-10-22-23-42-03-SRC-501-DST-84776114.wav|b") in
new stack
[Oct 22 23:42:03] -- Executing [...@macro-recordcall:9]
MacroExit("Local/84776...@outgoing_campaign-d2c3,2", "") in new stack
[Oct 22 23:42:03] -- Executing [84776...@outgoing_campaign:2]
Macro("Local/84776...@outgoing_campaign-d2c3,2",
"dialtrunk-failover|DAHDI/R1/4776114|DAHDI/R2/4776114|DAHDI/R3/4776114|DAHDI/R4/4776114|span_1|span_2|span_3|span_4")
in new stack
[Oct 22 23:42:03] == Begin MixMonitor Recording
Local/84776...@outgoing_campaign-d2c3,2
[Oct 22 23:42:03] -- Executing [...@macro-dialtrunk-failover:1]
GotoIf("Local/84776...@outgoing_campaign-d2c3,2", "0?1-fmsetcid|1") in new
stack
[Oct 22 23:42:03] -- Executing [...@macro-dialtrunk-failover:2]
GotoIf("Local/84776...@outgoing_campaign-d2c3,2", "0?1-setgbobname|1") in
new stack
[Oct 22 23:42:03] -- Executing [...@macro-dialtrunk-failover:3]
Set("Local/84776...@outgoing_campaign-d2c3,2", "CALLERID(num)=") in new
stack
[Oct 22 23:42:03] -- Executing [...@macro-dialtrunk-failover:4]
GotoIf("Local/84776...@outgoing_campaign-d2c3,2", "0?1-dial|1") in new
stack
[Oct 22 23:42:03] -- Executing [...@macro-dialtrunk-failover:5]
Set("Local/84776...@outgoing_campaign-d2c3,2", "CALLERID(all)=") in new
stack
[Oct 22 23:42:03] -- Executing [...@macro-dialtrunk-failover:6]
Goto("Local/84776...@outgoing_campaign-d2c3,2", "1-dial|1") in new stack
[Oct 22 23:42:03] -- Goto (macro-dialtrunk-failover,1-dial,1)
[Oct 22 23:42:03] -- Executing [1-d...@macro-dialtrunk-failover:1]
Dial("Local/84776...@outgoing_campaign-d2c3,2", "DAHDI/R1/4776114|90|tT")
in new stack
[Oct 22 23:42:03] -- Called R1/4776114
[Oct 22 23:42:05] -- DAHDI/4-1 answered
Local/84776...@outgoing_campaign-d2c3,2
[Oct 22 23:42:05]> Channel Local/84776...@outgoing_campaign-d2c3,1
was answered.
[Oct 22 23:42:05] -- Executing [...@queues:2]
Queue("Local/84776...@outgoing_campaign-d2c3,1", "501") in new stack
[Oct 22 23:42:05] -- Started music on hold, class 'default', on
channel 'Local/84776...@outgoing_campaign-d2c3,1'
[Oct 22 23:42:05] -- outgoing agentcall, to agent '10009', on
'Local/1...@default-4a9e,1'
[Oct 22 23:42:05] -- Executing [1...@default:1]
Dial("Local/1...@default-4a9e,2", "SIP/1000") in new stack
[Oct 22 23:42:05] -- Called 1000
[Oct 22 23:42:07] -- SIP/1000-0895df08 is ringing
[Oct 22 23:42:07] -- Agent/10009 is ringing
** PLease see las line with [Oct 22 23:42:05] when the output shows that
Called/ and then says DAHDI/4-1 answered.
[r...@pbx ~]# cat /asterisk/chan_dahdi.conf
[trunkgroups]
[channels]
language=ar
context=DID_trunk_1
signalling=fxs_ks
callwaiting=yes
hidecallerid=no
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=no
echocancelwhenbridged=no
relaxdtmf=yes
usedistinctiveringdetection=yes
usecallingpres=yes
busydetect=yes
callprogress=yes
rxgain=2.0
txgain=2.0
;
group=1
channel => 1-8
callgroup=1
pickupgroup=1
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Re: [asterisk-users] Cancel attended transfer
Agent 1 could park the call and have agent 2 pick it up from the lot. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Miguel Molina Sent: Monday, October 26, 2009 12:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Cancel attended transfer Hi folks, I have a simple question regarding attended transfers. I have some queues where agents take calls and I have configured attended transfers between queues. That is, the agent dials the attended transfer extension that routes it to the aproppiate transfer queue where the second agent answers and they both talk for a while. Finally the transferrer leaves the call with *, connecting the client, which was in hold, with the second agent. Is there a way for the transferrer to cancel the operation and return to the original call? If the call is queued and the agent leaves with *, it would end up in a blind transfer for the second agent who takes the transferred call. Is there a feature that can be configured to do that? I'm using asterisk 1.4.22 on a production server. TIA, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center ___ -- 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] What is the best way to configure this?
Hi, I have two servers ( A and B) in different towns. Both servers have pstn attached to them. Now I need to have calls coming to both servers to be answered on server A and then distributed between two sites. What is the best way of doing ? Having all calls to B forwarded to A on telco’s end? Having calls from B forwarded to A through internet? I tried to call forward the line on telco’s end but it still comes to the asterisk box and asterisk answers it generating clicking noise which I would like to avoid. How do I stop asterisk from answering that line? Thanks ___ -- 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] DAHDI not detecting RINGING Status on the Channel
It's not the DAHDI driver; it's the POTS service you are (presumably) using.
The DAHDI driver works fine with PRI/E1 interfaces, but POTS requires
"human" knowledge (it can't tell if a line is ringing/answered, etc). The
only "reasonable" solution I can suggest for this scenario is a
polarity/silence detect to keep you from processing many minutes of silence
and other garbage. LoopStart instead of KewlStart has been suggested in
some instances (possibly applicable to you since you are non-US).
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mariano
Lecuona
Sent: Monday, October 26, 2009 2:59 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] DAHDI not detecting RINGING Status on the Channel
I am using an 8 port tdm card and also I implemented a dialer using a
.call file generator. As you know on the .call you specify the channel to
call and then the contex/extension/priority to let dial plan continue when
the call is bridge.
My actual problem is that when the call process starts, asterisk (DAHDI)
sets the channel as answered when the truth is that on the other side the
channel has not started to ring yet. My felling is that the DAHDI driver
cannot detect /understand the signaling of the analog line. Here is the
evidence and configuration settings. I live in Argentina BTW
Console output.
[Oct 22 23:42:03] -- Attempting call on
Local/84776...@outgoing_campaign/n for 5...@queues:2 (Retry 1)
[Oct 22 23:42:03] -- Executing [84776...@outgoing_campaign:1]
Macro("Local/84776...@outgoing_campaign-d2c3,2",
"recordcall|501|84776114")in new stack
[Oct 22 23:42:03] -- Executing [...@macro-recordcall:1]
GotoIf("Local/84776...@outgoing_campaign-d2c3,2", "1?5:2") in new stack
[Oct 22 23:42:03] -- Goto (macro-recordcall,s,5)
[Oct 22 23:42:03] -- Executing [...@macro-recordcall:5]
Set("Local/84776...@outgoing_campaign-d2c3,2",
"FILEREC=2009-10-22-23-42-03-SRC-501-DST-84776114") in new stack
[Oct 22 23:42:03] -- Executing [...@macro-recordcall:6]
Set("Local/84776...@outgoing_campaign-d2c3,2",
"FILE_PATH=2009/10/22/2009-10-22-23-42-03-SRC-501-DST-84776114") in new
stack
[Oct 22 23:42:03] -- Executing [...@macro-recordcall:7]
Set("Local/84776...@outgoing_campaign-d2c3,2",
"CDR(userfield)=2009-10-22-23-42-03-SRC-501-DST-84776114.wav") in new
stack
[Oct 22 23:42:03] -- Executing [...@macro-recordcall:8]
MixMonitor("Local/84776...@outgoing_campaign-d2c3,2",
"/opt/rec/2009/10/22/2009-10-22-23-42-03-SRC-501-DST-84776114.wav|b") in
new stack
[Oct 22 23:42:03] -- Executing [...@macro-recordcall:9]
MacroExit("Local/84776...@outgoing_campaign-d2c3,2", "") in new stack
[Oct 22 23:42:03] -- Executing [84776...@outgoing_campaign:2]
Macro("Local/84776...@outgoing_campaign-d2c3,2",
"dialtrunk-failover|DAHDI/R1/4776114|DAHDI/R2/4776114|DAHDI/R3/4776114|DAHDI
/R4/4776114|span_1|span_2|span_3|span_4")
in new stack
[Oct 22 23:42:03] == Begin MixMonitor Recording
Local/84776...@outgoing_campaign-d2c3,2
[Oct 22 23:42:03] -- Executing [...@macro-dialtrunk-failover:1]
GotoIf("Local/84776...@outgoing_campaign-d2c3,2", "0?1-fmsetcid|1") in new
stack
[Oct 22 23:42:03] -- Executing [...@macro-dialtrunk-failover:2]
GotoIf("Local/84776...@outgoing_campaign-d2c3,2", "0?1-setgbobname|1") in
new stack
[Oct 22 23:42:03] -- Executing [...@macro-dialtrunk-failover:3]
Set("Local/84776...@outgoing_campaign-d2c3,2", "CALLERID(num)=") in new
stack
[Oct 22 23:42:03] -- Executing [...@macro-dialtrunk-failover:4]
GotoIf("Local/84776...@outgoing_campaign-d2c3,2", "0?1-dial|1") in new
stack
[Oct 22 23:42:03] -- Executing [...@macro-dialtrunk-failover:5]
Set("Local/84776...@outgoing_campaign-d2c3,2", "CALLERID(all)=") in new
stack
[Oct 22 23:42:03] -- Executing [...@macro-dialtrunk-failover:6]
Goto("Local/84776...@outgoing_campaign-d2c3,2", "1-dial|1") in new stack
[Oct 22 23:42:03] -- Goto (macro-dialtrunk-failover,1-dial,1)
[Oct 22 23:42:03] -- Executing [1-d...@macro-dialtrunk-failover:1]
Dial("Local/84776...@outgoing_campaign-d2c3,2", "DAHDI/R1/4776114|90|tT")
in new stack
[Oct 22 23:42:03] -- Called R1/4776114
[Oct 22 23:42:05] -- DAHDI/4-1 answered
Local/84776...@outgoing_campaign-d2c3,2
[Oct 22 23:42:05]> Channel Local/84776...@outgoing_campaign-d2c3,1
was answered.
[Oct 22 23:42:05] -- Executing [...@queues:2]
Queue("Local/84776...@outgoing_campaign-d2c3,1", "501") in new stack
[Oct 22 23:42:05] -- Started music on hold, class 'default', on
channel 'Local/84776...@outgoing_campaign-d2c3,1'
[Oct 22 23:42:05] -- outgoing agentcall, to agent '10009', on
'Local/1...@default-4a9e,1'
[Oct 22 23:42:05] -- Executing [1...@default:1]
Dial("Local/1...@default-4a9e,2", "SIP/1000") in new stack
[Oct 22 23:42:05] -- Called 1000
[Oct 22 23:42:07] -- SIP/1000-0895df08 is ringing
[Oct 22 23:42:07] -- Agent/10009 is ringing
** PLease see las line with [Oct 22 23:42:
Re: [asterisk-users] Answer call from another device
On Mon, 2009-10-26 at 14:58 -0500, Danny Nicholas wrote:
> *8 is the default value in features.conf to pick up a ringing line if you
> are in that ring group.
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Elliot Murdock
> Sent: Monday, October 26, 2009 2:52 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] Answer call from another device
>
> Hello!
>
> I remember a while back I saw a way to answer a call from a device
> that is not from the one ringing, but I don't remember what how to do
> it. Any help would be great!
One can take it even a little further than that depending on the phone.
We set up hints in extensions.conf so the programmable button lights on
our Snom phones indicate if a remote phone is ringing or off hook.
While it is ringing, a user can press the button which will issues the
*8 + extension and pick up the call. Here is an abbreviated example:
; Joe
exten => 613,hint,SIP/joe
; Mary
exten => 614,hint,SIP/mary
; Mike
exten => 616,hint,SIP/mike
; Enable call pickup for hinted stations from any possible source contexts
exten =>
_*8XXX,1,Pickup(${EXTEN:2...@a10pub&${EXTEN:2...@a10a&${EXTEN:2...@a10f)
We're pretty new to Asterisk so there may be a much better way but this
worked for us. Good luck - John
--
John A. Sullivan III
Open Source Development Corporation
+1 207-985-7880
jsulli...@opensourcedevel.com
http://www.spiritualoutreach.com
Making Christianity intelligible to secular society
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Re: [asterisk-users] What is the best way to configure this?
One suggestion - use "ex-girlfriend" logic on server b to only allow pickup of calls from Server A. - exten => s,1/5551212,Answer - exten => s,n,Hangup _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Robert Augustyn Sent: Monday, October 26, 2009 3:05 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] What is the best way to configure this? Hi, I have two servers ( A and B) in different towns. Both servers have pstn attached to them. Now I need to have calls coming to both servers to be answered on server A and then distributed between two sites. What is the best way of doing ? Having all calls to B forwarded to A on telco's end? Having calls from B forwarded to A through internet? I tried to call forward the line on telco's end but it still comes to the asterisk box and asterisk answers it generating clicking noise which I would like to avoid. How do I stop asterisk from answering that line? Thanks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AST-2009-007: ACL not respected on SIP INVITE
Asterisk Project Security Advisory - AST-2009-007 ++ | Product | Asterisk | |+---| | Summary | ACL not respected on SIP INVITE | |+---| | Nature of Advisory | Unauthorized calls allowed on prohibited networks | |+---| | Susceptibility | Remote unauthorized session | |+---| | Severity | Critical | |+---| | Exploits Known | No| |+---| |Reported On | October 18, 2009 | |+---| |Reported By | Thomas Athineou | |+---| | Posted On | October 26, 2009 | |+---| | Last Updated On | October 26, 2009 | |+---| | Advisory Contact | Jeff Peeler| |+---| | CVE Name | | ++ ++ | Description | A missing ACL check for handling SIP INVITEs allows a| | | device to make calls on networks intended to be | | | prohibited as defined by the "deny" and "permit" lines | | | in sip.conf. The ACL check for handling SIP | | | registrations was not affected. | ++ ++ | Resolution | Users should upgrade to a version listed in the | || "Corrected In" section below. | ++ ++ | Affected Versions| || |Product| Release Series | | |---++---| | Asterisk Open Source | 1.2.x | Unaffected| |---++---| | Asterisk Open Source | 1.4.x | Unaffected| |---++---| | Asterisk Open Source | 1.6.x | All 1.6.1 versions| |---++---| |Asterisk Addons| 1.2.x | Unaffected| |---++---| |Asterisk Addons| 1.4.x | Unaffected| |---++---| |Asterisk Addons| 1.6.x | Unaffected| |---++---| | Asterisk Business Edition | A.x.x | Unaffected| |---++---| | Asterisk Business Edition | B.x.x | Unaffected| |---++---| | Asterisk Business Edition | C.x.x | Unaffected| |---++---| | AsteriskNOW | 1.5 | Unaffected| |---++---| | s800i (Asterisk Appliance) | 1.2.x | Unaffected| ++ +--
Re: [asterisk-users] What is the best way to configure this?
I believe that for this, box B would have to answer the pstn line first …. I do not want that to happen Another question I have is how to configure Aastra phones to work with both servers and continue to work when internet connection is down? Thanks From:da...@debsinc.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Monday, October 26, 2009 4:12 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] What is the best way to configure this? One suggestion – use “ex-girlfriend” logic on server b to only allow pickup of calls from Server A. - exten => s,1/5551212,Answer - exten => s,n,Hangup From:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Robert Augustyn Sent: Monday, October 26, 2009 3:05 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] What is the best way to configure this? Hi, I have two servers ( A and B) in different towns. Both servers have pstn attached to them. Now I need to have calls coming to both servers to be answered on server A and then distributed between two sites. What is the best way of doing ? Having all calls to B forwarded to A on telco’s end? Having calls from B forwarded to A through internet? I tried to call forward the line on telco’s end but it still comes to the asterisk box and asterisk answers it generating clicking noise which I would like to avoid. How do I stop asterisk from answering that line? Thanks ___ -- 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] What is the best way to configure this?
In the example I gave you, B could only answer if the call was from box A (substitute 5551212 for the number for box A). _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Robert Augustyn Sent: Monday, October 26, 2009 3:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] What is the best way to configure this? I believe that for this, box B would have to answer the pstn line first .. I do not want that to happen Another question I have is how to configure Aastra phones to work with both servers and continue to work when internet connection is down? Thanks From: da...@debsinc.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Monday, October 26, 2009 4:12 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] What is the best way to configure this? One suggestion - use "ex-girlfriend" logic on server b to only allow pickup of calls from Server A. - exten => s,1/5551212,Answer - exten => s,n,Hangup _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Robert Augustyn Sent: Monday, October 26, 2009 3:05 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] What is the best way to configure this? Hi, I have two servers ( A and B) in different towns. Both servers have pstn attached to them. Now I need to have calls coming to both servers to be answered on server A and then distributed between two sites. What is the best way of doing ? Having all calls to B forwarded to A on telco's end? Having calls from B forwarded to A through internet? I tried to call forward the line on telco's end but it still comes to the asterisk box and asterisk answers it generating clicking noise which I would like to avoid. How do I stop asterisk from answering that line? Thanks ___ -- 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] What is the best way to configure this?
For question #2, configure one line on each phone to A and one to B. If you did an IAX2 connection from A to B, you could channel calls from A to B that way and do PSTN answering on B if the internet connection was not up (IAX2 show peers). _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Robert Augustyn Sent: Monday, October 26, 2009 3:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] What is the best way to configure this? I believe that for this, box B would have to answer the pstn line first .. I do not want that to happen Another question I have is how to configure Aastra phones to work with both servers and continue to work when internet connection is down? Thanks From: da...@debsinc.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Monday, October 26, 2009 4:12 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] What is the best way to configure this? One suggestion - use "ex-girlfriend" logic on server b to only allow pickup of calls from Server A. - exten => s,1/5551212,Answer - exten => s,n,Hangup _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Robert Augustyn Sent: Monday, October 26, 2009 3:05 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] What is the best way to configure this? Hi, I have two servers ( A and B) in different towns. Both servers have pstn attached to them. Now I need to have calls coming to both servers to be answered on server A and then distributed between two sites. What is the best way of doing ? Having all calls to B forwarded to A on telco's end? Having calls from B forwarded to A through internet? I tried to call forward the line on telco's end but it still comes to the asterisk box and asterisk answers it generating clicking noise which I would like to avoid. How do I stop asterisk from answering that line? Thanks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.6.1.8 Now Available
The Asterisk Development Team has announced the releases of Asterisk 1.6.1.8. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of 1.6.1.8 resolves an issue where an ACL check is not present for verifying SIP INVITEs. For more information about the details of this vulnerability, please read the security advisory AST-2009-007, which was released at the same time as this announcement. The Asterisk 1.6.1 series is the only fully released version which contains this vulnerability. Releases from previous branches (1.6.0, 1.4, 1.2) are not affected. In addition, Asterisk users may notice that we skipped the version number 1.6.1.7. This was intentional, in an effort to avoid confusion about what a particular release contains. Asterisk 1.6.1.7 had candidates for release made, so backtracking on those changes in a release with the same version number might be confusing. The next release candidate, which would have been 1.6.1.7-rc3, will be released with additional changes as 1.6.1.9-rc1. For a full list of changes in the current release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.1.8 Release announcement AST-2009-007 is available at: http://downloads.asterisk.org/pub/security/AST-2009-007.pdf Thank you for your continued support of Asterisk! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DAHDI not detecting RINGING Status on the Channel
For some reason I am not able to set loopstart instead of kewlstart:
Console out put:
[Oct 26 20:58:40] == Parsing '/etc/asterisk/chan_dahdi.conf': [Oct 26
20:58:40] Found
[Oct 26 20:58:40] -- Registered channel 1, FXS Loopstart signalling
[Oct 26 20:58:40] -- Registered channel 2, FXS Loopstart signalling
[Oct 26 20:58:40] ERROR[23050]: chan_dahdi.c:7677 mkintf: Signalling
requested on channel 3 is FXS Loopstart but line is in FXS Kewlstart
signalling
[Oct 26 20:58:40] ERROR[23050]: chan_dahdi.c:11294 build_channels: Unable to
register channel '1-8'
pbx*CLI> module load chan_dahdi.so
any ideas? I am still tring to set some other parameters like:
cidsignalling=
cidstart=
Thanks
ML
2009/10/26 Danny Nicholas
> It’s not the DAHDI driver; it’s the POTS service you are (presumably)
> using. The DAHDI driver works fine with PRI/E1 interfaces, but POTS
> requires “human” knowledge (it can’t tell if a line is ringing/answered,
> etc). The only “reasonable” solution I can suggest for this scenario is a
> polarity/silence detect to keep you from processing many minutes of silence
> and other garbage. LoopStart instead of KewlStart has been suggested in
> some instances (possibly applicable to you since you are non-US).
>
>
> --
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Mariano Lecuona
> *Sent:* Monday, October 26, 2009 2:59 PM
>
> *To:* asterisk-users@lists.digium.com
> *Subject:* [asterisk-users] DAHDI not detecting RINGING Status on the
> Channel
>
>
>
> I am using an 8 port tdm card and also I implemented a dialer using a
> .call file generator. As you know on the .call you specify the channel to
> call and then the contex/extension/priority to let dial plan continue when
> the call is bridge.
> My actual problem is that when the call process starts, asterisk (DAHDI)
> sets the channel as answered when the truth is that on the other side the
> channel has not started to ring yet. My felling is that the DAHDI driver
> cannot detect /understand the signaling of the analog line. Here is the
> evidence and configuration settings. I live in Argentina BTW
>
> Console output.
> [Oct 22 23:42:03] -- Attempting call on
> Local/84776...@outgoing_campaign/n for 5...@queues:2 (Retry 1)
> [Oct 22 23:42:03] -- Executing [84776...@outgoing_campaign:1]
> Macro("Local/84776...@outgoing_campaign-d2c3,2",
> "recordcall|501|84776114")in new stack
> [Oct 22 23:42:03] -- Executing [...@macro-recordcall:1]
> GotoIf("Local/84776...@outgoing_campaign-d2c3,2", "1?5:2") in new stack
> [Oct 22 23:42:03] -- Goto (macro-recordcall,s,5)
> [Oct 22 23:42:03] -- Executing [...@macro-recordcall:5]
> Set("Local/84776...@outgoing_campaign-d2c3,2",
> "FILEREC=2009-10-22-23-42-03-SRC-501-DST-84776114") in new stack
> [Oct 22 23:42:03] -- Executing [...@macro-recordcall:6]
> Set("Local/84776...@outgoing_campaign-d2c3,2",
> "FILE_PATH=2009/10/22/2009-10-22-23-42-03-SRC-501-DST-84776114") in new
> stack
> [Oct 22 23:42:03] -- Executing [...@macro-recordcall:7]
> Set("Local/84776...@outgoing_campaign-d2c3,2",
> "CDR(userfield)=2009-10-22-23-42-03-SRC-501-DST-84776114.wav") in new
> stack
> [Oct 22 23:42:03] -- Executing [...@macro-recordcall:8]
> MixMonitor("Local/84776...@outgoing_campaign-d2c3,2",
> "/opt/rec/2009/10/22/2009-10-22-23-42-03-SRC-501-DST-84776114.wav|b") in
> new stack
> [Oct 22 23:42:03] -- Executing [...@macro-recordcall:9]
> MacroExit("Local/84776...@outgoing_campaign-d2c3,2", "") in new stack
> [Oct 22 23:42:03] -- Executing [84776...@outgoing_campaign:2]
> Macro("Local/84776...@outgoing_campaign-d2c3,2",
>
> "dialtrunk-failover|DAHDI/R1/4776114|DAHDI/R2/4776114|DAHDI/R3/4776114|DAHDI/R4/4776114|span_1|span_2|span_3|span_4")
> in new stack
> [Oct 22 23:42:03] == Begin MixMonitor Recording
> Local/84776...@outgoing_campaign-d2c3,2
> [Oct 22 23:42:03] -- Executing [...@macro-dialtrunk-failover:1]
> GotoIf("Local/84776...@outgoing_campaign-d2c3,2", "0?1-fmsetcid|1") in new
> stack
> [Oct 22 23:42:03] -- Executing [...@macro-dialtrunk-failover:2]
> GotoIf("Local/84776...@outgoing_campaign-d2c3,2", "0?1-setgbobname|1") in
> new stack
> [Oct 22 23:42:03] -- Executing [...@macro-dialtrunk-failover:3]
> Set("Local/84776...@outgoing_campaign-d2c3,2", "CALLERID(num)=") in new
> stack
> [Oct 22 23:42:03] -- Executing [...@macro-dialtrunk-failover:4]
> GotoIf("Local/84776...@outgoing_campaign-d2c3,2", "0?1-dial|1") in new
> stack
> [Oct 22 23:42:03] -- Executing [...@macro-dialtrunk-failover:5]
> Set("Local/84776...@outgoing_campaign-d2c3,2", "CALLERID(all)=") in new
> stack
> [Oct 22 23:42:03] -- Executing [...@macro-dialtrunk-failover:6]
> Goto("Local/84776...@outgoing_campaign-d2c3,2", "1-dial|1") in new stack
> [Oct 22 23:42:03] -- Goto (macro-dialtrunk-failover,1-dial,1)
> [Oct 22 23:42:03] -- Executing [1-d...@macro-dialtrunk-failover:1]
>
Re: [asterisk-users] No tone, one way communcation.
my lsdahdi output is; 1. [r...@elastix ~]# lsdahdi ### Span 1: WCTDM/8 "YSTDM8xx REV E Board 9" (MASTER) 1 FXSFXOKS (In use) 2 FXSFXOKS (In use) 3 EMPTY 4 FXSFXOKS (In use) 5 FXOFXSKS (In use) RED 6 FXOFXSKS (In use) RED 7 FXOFXSKS (In use) RED 8 FXOFXSKS (In use) RED seems fxs port 3 is disconnected but i will fix it. The analog phone is connected to fxs port 1. The dahdi_hardware output is; 2. [r...@elastix ~]# dahdi_hardware pci::04:02.0 ystdm8xx+e159:0001 Yeastar YSTDM8xx From: Tzafrir Cohen To: asterisk-users@lists.digium.com Sent: Mon, October 26, 2009 4:05:16 PM Subject: Re: [asterisk-users] No tone, one way communcation. On Mon, Oct 26, 2009 at 05:02:18AM -0700, PATRICK KANGETHE wrote: > 1. When i connected my analog phone to fxs card, i cannot get dial tone what > could be the problem? What is the output of: lsdahdi dahdi_hardware > > I am using elastix 1.5.2 based on centos 5.2 Final. Consider also asking there. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- 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
