Re: [asterisk-users] Trunk SIP and configuration
thank you for your reply. I'm French.
I added the field operator, and nothing.
when I call sda, it does not work.
I bought numbers sda.
I have a voip access.
Does the operator configure sip accounts?
Does the operator configures the corresponding sip sda?
Regards.
Ludo
2009/4/1 Carlos Rojas crt.ro...@gmail.com
Hello,
I don't speak english very well but i think.
[operador]
qualify=yes
nat=yes
host=192.168.700.50
insecure=invite,port
canreinvite=no
context=default
disallow=all
allow=ulaw
allow=g729
in your extensions.conf
exten = _00X,1, Dial (SIP/operador/${EXTEN},60,tT)
Best Regards
Carlos Rojas
On Wed, Apr 1, 2009 at 10:45 AM, ludo perrot ludoper...@gmail.com wrote:
hello,
I am beginning to asterisk.
I have a sip trunk access to operator and VPN access with operator.
i booked 10 sda numbers.
IP adress asterisk : 192.168.600.1
IP adress operator : 192.168.700.50
i can ping on 192.168.700.50
# cat sip.conf
[general]
context=default
srvlookup=yes
port = 5060
disallow=all
allow=gsm
allow=alaw
allow=ulaw
[1000]
username=1000
type=friend
qualify=yes
secret=3615
nat=no
host=192.168.600.3
canreinvite=no
context=appels_entrants
[Catherine]
usename=1010
type=friend
qualify=yes
secret=5768
nat=yes
host=192.168.600.4
canreinvite=yes
context=default
disallow=all
allow=ulaw
# extensions.conf
exten = _00X,1, Dial (SIP/192.168.700.50/${EXTEN})
How do I configure IP operator ?
I have 10 numbers sda. Where do I configure sda numbers ?
Thanks.
Ludovic
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Re: [asterisk-users] Extract a MOS value from Asterisk CDR
Indeed, we already have
- the function to convert R factor to MOS
- the R function R = R0 -Is-Id-Ie+A
- the codec used
- the rtt, rx/tx jitter, packet loss
What ye do not have but is needed:
- A factor, a note between 0 and 20 - 0 for landlines
- the Burst Ratio, I'm using 1 (random repartition)
I already have an openoffice calc function to calculate the MOS regarding the
rtt, packet loss, codec, I have to add the jitter!
Here are the URL I have used
* http://www.itu.int/rec/T-REC-G.107-200503-S/en
* http://www.ixiacom.com/library/white_papers/display?skey=voip_quality
* http://www.itu.int/ITU-T/studygroups/com12/emodelv1/tut.htm
Have a nice day!
-- --
Marc LEURENT
Ingénieur VoIP
DECKPOINT SA
Une société du groupe VTX Telecom
Rue Eugène-Marziano 15 - 1227 Les Acacias
http://www.vtx.ch - marc.leur...@vtx-telecom.ch
VTX, votre partenaire telecom proche de vous !
Le Thursday 02 April 2009 03.27:48 John Todd, vous avez écrit :
Thank you for the interesting links on MOS values and calculations!
It seems that many (most?) of the values that are used to construct R
and MOS could be obtained from the data that exists within the
dialplan, at least as far as the visible RTP path is concerned. Or
is there data missing in the current RTCP statistics that would be
required to make correct R/MOS value estimates? (If so, then that's
on-topic for asterisk-dev, otherwise this should be moved to asterisk-
users...)
Here is the data that I think is already visible:
- codec choices
- round-trip delay to RTP endpoint
- packet loss
- jitter
I think it is too complex to determine Irecency, A or packet loss
bursts unless there is significant additional code added to Asterisk
to capture more granular time-slices of data on each call. I also
think that mid-call codec changes should not be considered due to
complexity. Currently, I think this is un-necessary since most people
don't even seem to compute MOS to start with.
So in your examination you may come up with a script or dialplan that
creates a synthetic R or MOS value - could you post it to a blog, or
if it is very short, to the asterisk-users mailing list? I think this
would be worthwhile.
JT
On Apr 1, 2009, at 2:57 PM, Mindaugas Kezys wrote:
Sorry for replying for the second time, but this issue is
interesting for me
also.
I found such link: http://www.nessoft.com/kb/50
And this:
http://www.jdsu.com/product-literature/voipstats_an_acc_tm_ae.pdf
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
VoIP Billing and Routing Solutions
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Marc
Leurent
Sent: 2009 m. balandžio 1 d. 18:15
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Extract a MOS value from Asterisk CDR
Hello all,
I'm tring to retrieve a formula to calculate a MOS value from
Asterisk RTCP
stats...
Have you got any idea how to do it?
Thanks
I'm reading all G.107 ITU docs to retrieve something...
I'm saving the SIP RTCP stats with:
[macro-hangupcall]
exten = s,1,Set(CDR(userfield)=${CHANNEL(rtpqos|audio|all)})
exten = s,n,ResetCDR(vw)
exten = s,n,NoCDR()
So I retrieve these values in my MySQL CDR table in order to
calculate a MOS
value:
ssrc
=
592614191;themssrc=0;lp=1;rxjitter=0.00;rxcount=0;txjitter=0.0
0;txcount=20734;rlp=0;rtt=0.094000
codec used: g711a
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lf...@leurent.eu
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Digium, Inc. | Asterisk Open Source Community Director
445 Jan Davis Drive NW - Huntsville AL 35806 - USA
direct: +1-256-428-6083 http://www.digium.com/
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Re: [asterisk-users] 400 calls at g711 how much cpu power
On Wed, 1 Apr 2009, Erick Perez wrote: We are planning to run an outbound only campaign. A 20-second voice message will be played to callers and our dialer on machine1 will send to machine2-asterisk (1.4) instructions to dial 400 calls, play the message and hang up. This will be done for about 1 million phones. The asterisk box will communicate via SIP to a voice carrier. the voice carrier will then place the calls on pstn. The codec will be g711. So we will never do any transcoding. I have been calculating the CPU power required to do the calls and in previous posting the usual calculation is about 40MHZ per leg when no transcoding is involved. So if we use the 40MHZ rule, we are talking about 40*400=16000MHZ or 1.6Ghz. Comments? I don't personally think CPU GHz is a good measure for something like this, there are many other factors at work when things get big... One thing I'd be concerend about is the number of packets per second and how the underlying hardware is going to cope with shoving them out - and remember VoIP is bi-directional, so even if you're just sending data out, there will still be data coming in at the same rate... So 50 packets (of 160 bytes + IP overhead) per second, each way times 400 is 40,000 packets per second that the system has to get to and from the Ethernet card. You might want to check the specification of your router too to make sure it can handle that load... Oh, and bandwidth - you're looking at 80Kb/sec for each call - that's going to need 32,000Kb/sec or 32Mb/sec - and remember that's each way.. As for the server - get *everything* in RAM. At least with no disk IO, it's one less thing going over the PCI bus when it's running - even then, you may want to look for a server motherboard with multiple PCI buses, although working that out beforehand is sometimes problematic unless you have the time to go through the motherboard manuals in detail, or know beforehand what motherboard does what... And you may find that a uni-processor server is better than multi-core too to minimise locks at the kernel level with multiple cores accessing the same Ethernet hardware... And you can always use 2, 3 or 4, etc. outbound call servers - with the one dialler round-robbining the calls to each server. That might be a better idea anyway than one big beast of a server. Good luck! (And let us know how you get on!) Gordon ___ -- 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] [CLOSED] Asterisk + OpenSIPs Integration - Rewrite URI on Trunk Numbers of a SIP Trunk
Hello, all.
This is just an email to inform you I have added a SIP header in Asterisk SIP
message that is handled by the proxy:
On Asterisk extensions.conf:
SIPAddHeader(X-number-to-dial: ${NUMBERTOREACH})
Dial(SIP/${MAINPEER}|100|t)
and on OpenSIPS:
if (is_present_hf(X-number-to-dial)) {
xlog(L_DBG, GOING TO replace URI username with X-number-to-dial\r\n);
xlog(L_DBG, Print $(hdr(X-number-to-dial)) \r\n);
subst_user('/(.*)/$(hdr(X-number-to-dial))/');# Substitute the URI
phone number with the one in X-number-to-dial SIP Header
subst('/^(To|t):(.*)sip:[...@]*@(.*)$/\1:\2sip:
$(hdr(X-number-to-dial))@\3/ig');
}
Have a nice day!
-- --
Marc LEURENT
Le Monday 23 March 2009 13.41:59 Marc Leurent, vous avez écrit :
I have spoken to quickly,
Usually Asterisk on an incoming call sends an INVITE
Reg.Contact
Number@Reg Contact IP to the Peer IP. With the command you gave me, it
is possible to send anINVITE othernumber@Peer IP to the
Peer IP.
What I would like to do is to sendINVITE
othernumber@Reg Contact IP
to the Peer IP in order for the request to be forwarded by the proxy!
Is it possible to do something like:
Dial(SIP/sip:1...@192.168.10.125:5060@1003 )
in Order to send INVITE 1...@1005 IP to 1003 device IP
Thanks!
Le Monday 23 March 2009 12.03:55 Marc Leurent, vous avez écrit :
Thank you, this is exactly what I needed!!
In order to Dial any number to a registered peer, I just have to enter
Dial(SIP/anynum...@sippeername) Best Regards!
Le Monday 23 March 2009 11.31:31 Alex Balashov, vous avez écrit :
The Request URI generated in an INVITE originated by Asterisk is
governed entirely by the parameters passed to Dial().
For example:
Dial(SIP/1...@peer_name)
... will generate a Request URI of
1...@host.or.ip.of.sip.conf.peer.named.peer_name.
It is also possible to send requests to hosts that are not explicitly
defined in sip.conf, with the caveat that only background [general]
sip.conf settings will then apply:
Dial(SIP/1...@ip.of.peer.not.in.sip.conf)
Marc Leurent wrote:
Hello,
it is not an OpenSIPs problem I have, it's an Asterisk one,
I would like to change the URI in message generated by Asterisk.
Thanks
Le Monday 23 March 2009 10.35:09 Alex Balashov, vous avez écrit :
Modify the $ru pseudovariable or use rewritehostport() out of core.
This is not the right mailing list. This belongs on the
OpenSIPS/OpenSER lists.
There is also a mailing list we operate called
SER-Asterisk-Interwork that is specifically intended to address SER*
/ Asterisk integration issues:
http://lists.evaristesys.com/mailman/listinfo/ser-asterisk-interwork
* Anything from the [Open]SER family.
lftsy wrote:
Hye everybody, anyone has any idea how to help me?
To resume, I just want to know how to change the IP in the URI sent
by Asterisk (first line of SIP packets)
Thanks for your time!
++
On Fri, 20 Mar 2009 15:09:55 +0100, Marc Leurent lf...@leurent.eu
wrote:
Hello All,
I have a little complicated question about the Dial command.
I use OpenSIPs to loadbalance Asterisk Servers, and Users are
registered on Asterisk servers.
Asterisk use the Reg. Contact entry to reach the UAC via the
OpenSIPs server. Everything works except for trunk numbers:
For each peer on Asterisk, Addr-IP is IP of the Proxy and Reg.
Contact is the IP where the proxy will relay the packet to reach
the
UAC.
Ex: with a trunk 0123400010 - 0123400019 with 0123400010 as the
sip
peer.
When a number from a trunk is called, like 0123400019 the Reg.
Contact of the main number is not used.
For the time being, I use Dial(SIP/0123400010/0123400019) but it
It sends an
INVITE sip:0123400...@proxyip to the proxy
whereas it should send
INVITE sip:0123400019@Reg. Contact of the main number to the
proxy
So I'm trying use the Dial Command with
Dial(SIP/0123400010/0123400019@Reg. Contact of the main number)
but it doesn't work
Have you got any idea how to rewrite the IP of the URI sent?
Thanks!
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lf...@leurent.eu
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Re: [asterisk-users] PRI problem
I had the exact same problem and errors some time ago (search the archives for PRI dropping #2) using Asterisk 1.4.18, Zaptel and a Digium TE121. I tried all kind of things, had telco technicians come out and whatnot. The solution was two-folded - 1) I reinstalled my server, 2) I updated to Asterisk 1.4.24, replaced Zaptel with latest DAHDI. In the DAHDI case I even had to use latest Subversion revision due to some bug (but that was related to the TE121-cards I think). Since then I haven't had any issues at all, so consider updating Asterisk and Zaptel-DAHDI 2009/3/31 Steven J. Douglas stev...@moij.biz: Hi Brandon, When using the current straight cable, it sometimes worked i.e. I can make calls from the PSTN into the asterisk. Do you still think that I should try a crossover cable? Thanks. Regards, Steve. Brandon B. wrote: Try a T1 crossover cable: http://www.voip-info.org/wiki/view/crossover+T1+cable On Tue, Mar 31, 2009 at 12:37 AM, Steven J. Douglas stev...@moij.biz mailto:stev...@moij.biz wrote: Hi guys, I've been trying to get my ISDN-10 line up for the past few days, but its been going up and down. I am using OpenVox D110P card on asterisk version 1.4.21. It seems to me like a cable problem. I tried using Ethernet straight cable (12, 45, 36, 78) and also a straight cable where the twisted pairs are on 12, 34, 56 and 78. The problem remains the same. /*etc/zaptel.conf* loadzone=sg defaultzone=sg # PRI Span span=1,1,0,ccs,hdb3,crc4 bchan=1-15 dchan=16 bchan=17-31 */etc/asterisk/zapata.conf* language=en progzone=sg musiconhold=default ; PRI Set Up context=inbound-pri1 switchtype=euroisdn signalling=pri_cpe pridialplan=national overlapdial=yes immediate=no faxdetect=both overlapdial=no usecallerid=yes usecallingpres=yes callerid=asreceived group=9 channel = 1-15 channel = 17-31 The following are the messages that keep repeating. == Primary D-Channel on span 1 down Mar 31 14:34:05 WARNING[2361]: chan_zap.c:2682 pri_find_dchan: No D-channels available! Using Primary channel 16 as D-channel anyway! Mar 31 14:34:07 NOTICE[2362]: chan_zap.c:6880 handle_init_event: Alarm cleared on channel 1 Mar 31 14:34:07 NOTICE[2362]: chan_zap.c:6880 handle_init_event: Alarm cleared on channel 2 Mar 31 14:34:07 NOTICE[2362]: chan_zap.c:6880 handle_init_event: Alarm cleared on channel 3 Mar 31 14:34:07 NOTICE[2362]: chan_zap.c:6880 handle_init_event: Alarm cleared on channel 4 Mar 31 14:34:07 NOTICE[2362]: chan_zap.c:6880 handle_init_event: Alarm cleared on channel 5 Mar 31 14:34:07 NOTICE[2362]: chan_zap.c:6880 handle_init_event: Alarm cleared on channel 6 Mar 31 14:34:07 NOTICE[2362]: chan_zap.c:6880 handle_init_event: Alarm cleared on channel 7 Mar 31 14:34:07 NOTICE[2362]: chan_zap.c:6880 handle_init_event: Alarm cleared on channel 8 Mar 31 14:34:07 NOTICE[2362]: chan_zap.c:6880 handle_init_event: Alarm cleared on channel 9 Mar 31 14:34:07 NOTICE[2362]: chan_zap.c:6880 handle_init_event: Alarm cleared on channel 10 Mar 31 14:34:07 NOTICE[2362]: chan_zap.c:6880 handle_init_event: Alarm cleared on channel 11 Mar 31 14:34:07 NOTICE[2362]: chan_zap.c:6880 handle_init_event: Alarm cleared on channel 12 Mar 31 14:34:07 NOTICE[2362]: chan_zap.c:6880 handle_init_event: Alarm cleared on channel 13 Mar 31 14:34:07 NOTICE[2362]: chan_zap.c:6880 handle_init_event: Alarm cleared on channel 14 Mar 31 14:34:07 NOTICE[2362]: chan_zap.c:6880 handle_init_event: Alarm cleared on channel 15 Mar 31 14:34:07 NOTICE[2362]: chan_zap.c:6880 handle_init_event: Alarm cleared on channel 17 Mar 31 14:34:07 NOTICE[2362]: chan_zap.c:6880 handle_init_event: Alarm cleared on channel 18 Mar 31 14:34:07 NOTICE[2362]: chan_zap.c:6880 handle_init_event: Alarm cleared on channel 19 Mar 31 14:34:07 NOTICE[2362]: chan_zap.c:6880 handle_init_event: Alarm cleared on channel 20 Mar 31 14:34:07 NOTICE[2362]: chan_zap.c:6880 handle_init_event: Alarm cleared on channel 21 Mar 31 14:34:07 NOTICE[2362]: chan_zap.c:6880 handle_init_event: Alarm cleared on channel 22 Mar 31 14:34:07 NOTICE[2362]: chan_zap.c:6880 handle_init_event: Alarm cleared on channel 23 Mar 31 14:34:07 NOTICE[2362]: chan_zap.c:6880 handle_init_event: Alarm cleared on channel 24 Mar 31 14:34:07 NOTICE[2362]: chan_zap.c:6880 handle_init_event: Alarm cleared on channel 25 Mar 31 14:34:07 NOTICE[2362]: chan_zap.c:6880 handle_init_event: Alarm cleared on channel 26 Mar 31 14:34:07 NOTICE[2362]: chan_zap.c:6880 handle_init_event: Alarm cleared on channel 27 Mar 31 14:34:07 NOTICE[2362]: chan_zap.c:6880 handle_init_event:
Re: [asterisk-users] [Closed] no ringtone - just silence/bridging ofexternal calls
Hi! I used a work around to the problem. I added a Playback(silence/1) quite after the Answer() and now everything is working fine again. 100, 1, Answer() 100, 2, Playback(silence/1) 100, 3, Dial(SIP/XX,,r) Hope this helps, Alex Alex Mosburger -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jean-Michel Hiver Sent: Montag, 30. März 2009 16:51 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] no ringtone - just silence/bridging ofexternal calls Hello For the ringtone try progressinband=yes in sip.conf. I don't think you can bridge do a ringback at the same time, why not proxy the RTP and send the ringback yourself using the 'm' modifier? Cheers Jean-Michel. 2009/3/30, alex.mosbur...@orange-ftgroup.com alex.mosbur...@orange-ftgroup.com: Hi Community! If this issue was already topic, please excuse or delete my request... Topic 1 no ringtone: I configured a SIP registration with my SIP provider (SIPCall). Everything works fine except the ring tone for the caller. The caller hears silence until the called party takes up the phone. I used the DIAL command with the r and R option but no luck... :( Has anybody the same problem than me and a resolution for it? - Topic 2 external bridging: The prior approach was to bridge to external calls. An external SIP number terminates and will be re-routed back to a mobile phone number. The session was first packet2packet switched, which did not work. After setting reinvite=yes, the bridge works. Now I added 2 internal extensions to the mobile phone number in the DIAL command -- did not work (mobile phone rings but no communication possible; just silence). Topology: SIP Provider -- Asterisk -- SIP Provider -- Mobile phone /- ext 10 /- ext 20 The DIAL command was: Dial(SIP/06544564...@sipcall.atSIP/10SIP/20,,r) The aim is that all extensions and the mobile rings and the first pick up takes the call. During call setup music on hold would be good... It shows no errors in the debug of the CLI. I would appreciate if somebody could help me. Thanks, Alex * This message and any attachments (the message) are confidential and intended solely for the addressees. Any unauthorised use or dissemination is prohibited. Messages are susceptible to alteration. France Telecom Group shall not be liable for the message if altered, changed or falsified. If you are not the intended addressee of this message, please cancel it immediately and inform the sender. ___ -- 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 -- Jean-Michel Hiver - Synapse co-founder CTO GSM +262 692 828 070 ___ -- 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 * This message and any attachments (the message) are confidential and intended solely for the addressees. Any unauthorised use or dissemination is prohibited. Messages are susceptible to alteration. France Telecom Group shall not be liable for the message if altered, changed or falsified. If you are not the intended addressee of this message, please cancel it immediately and inform the sender. ___ -- 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] async agi question
Hi Henrik,
I would like to do the same thing you are doing here. I want to implement an
external queue functionality so I need to stop a play file launched previously
with an async agi command on caller's channel, sending the call to agent's
extension.
I'm redirecting caller's channel with REDIRECT while playing is taking place
but I'm always getting a hang up on caller's channel.
I'm using:
asterisk-1.4.18
asterisk-addons-1.4.7
async agi patch 2007-12-11 10:34:12 (the one back-ported to 1.4)
Both caller and agent are using 501 and 500 extensions and the async agi loop
is waiting on 800, for example. The caller is dialing 800 where a play file is
commanded through and async agi stream file command by the application.
The relevant part of extensions.conf follows:
exten = _5.,1,Noop(SIP call on 'sip_sercom' a ${EXTEN});
exten = _5.,n,Wait(1);
exten = _5.,n,Dial(SIP/${EXTEN},${TIMEOUTDIAL},Tto);
exten = _5.,n,Hangup();
exten = _8.,1,Noop(every thing starting 8 ${EXTEN});
exten = _8.,n,AGI(agi:async);
exten = _8.,n,Hangup();
And the redirect command the application is sending to is:
Action: Redirect
Channel: SIP/501-081f0730
Exten: 500
Context: sip_sercom
Priority: 1
Therefore, Henrik, could you show me your related dial plan and the redirect
command you are sending? I wasn't able to see what I'm getting wrong.
thanks in advanced
Jose M Arias
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Re: [asterisk-users] Extract a MOS value from Asterisk CDR
Could you share with us your Openoffice callc function?
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
VoIP Billing and Routing Solutions
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Marc Leurent
Sent: 2009 m. balandžio 2 d. 11:29
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Extract a MOS value from Asterisk CDR
Indeed, we already have
- the function to convert R factor to MOS
- the R function R = R0 -Is-Id-Ie+A
- the codec used
- the rtt, rx/tx jitter, packet loss
What ye do not have but is needed:
- A factor, a note between 0 and 20 - 0 for landlines
- the Burst Ratio, I'm using 1 (random repartition)
I already have an openoffice calc function to calculate the MOS regarding the
rtt, packet loss, codec, I have to add the jitter!
Here are the URL I have used
* http://www.itu.int/rec/T-REC-G.107-200503-S/en
* http://www.ixiacom.com/library/white_papers/display?skey=voip_quality
* http://www.itu.int/ITU-T/studygroups/com12/emodelv1/tut.htm
Have a nice day!
-- --
Marc LEURENT
Ingénieur VoIP
DECKPOINT SA
Une société du groupe VTX Telecom
Rue Eugène-Marziano 15 - 1227 Les Acacias
http://www.vtx.ch - marc.leur...@vtx-telecom.ch
VTX, votre partenaire telecom proche de vous !
Le Thursday 02 April 2009 03.27:48 John Todd, vous avez écrit :
Thank you for the interesting links on MOS values and calculations!
It seems that many (most?) of the values that are used to construct R
and MOS could be obtained from the data that exists within the
dialplan, at least as far as the visible RTP path is concerned. Or
is there data missing in the current RTCP statistics that would be
required to make correct R/MOS value estimates? (If so, then that's
on-topic for asterisk-dev, otherwise this should be moved to asterisk-
users...)
Here is the data that I think is already visible:
- codec choices
- round-trip delay to RTP endpoint
- packet loss
- jitter
I think it is too complex to determine Irecency, A or packet loss
bursts unless there is significant additional code added to Asterisk
to capture more granular time-slices of data on each call. I also
think that mid-call codec changes should not be considered due to
complexity. Currently, I think this is un-necessary since most people
don't even seem to compute MOS to start with.
So in your examination you may come up with a script or dialplan that
creates a synthetic R or MOS value - could you post it to a blog, or
if it is very short, to the asterisk-users mailing list? I think this
would be worthwhile.
JT
On Apr 1, 2009, at 2:57 PM, Mindaugas Kezys wrote:
Sorry for replying for the second time, but this issue is
interesting for me
also.
I found such link: http://www.nessoft.com/kb/50
And this:
http://www.jdsu.com/product-literature/voipstats_an_acc_tm_ae.pdf
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
VoIP Billing and Routing Solutions
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Marc
Leurent
Sent: 2009 m. balandžio 1 d. 18:15
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Extract a MOS value from Asterisk CDR
Hello all,
I'm tring to retrieve a formula to calculate a MOS value from
Asterisk RTCP
stats...
Have you got any idea how to do it?
Thanks
I'm reading all G.107 ITU docs to retrieve something...
I'm saving the SIP RTCP stats with:
[macro-hangupcall]
exten = s,1,Set(CDR(userfield)=${CHANNEL(rtpqos|audio|all)})
exten = s,n,ResetCDR(vw)
exten = s,n,NoCDR()
So I retrieve these values in my MySQL CDR table in order to
calculate a MOS
value:
ssrc
=
592614191;themssrc=0;lp=1;rxjitter=0.00;rxcount=0;txjitter=0.0
0;txcount=20734;rlp=0;rtt=0.094000
codec used: g711a
--
-- --
Marc LEURENT
lf...@leurent.eu
___
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---
John Todd email:jt...@digium.com
Digium, Inc. | Asterisk Open Source Community Director
445 Jan Davis Drive NW - Huntsville AL 35806 - USA
direct: +1-256-428-6083 http://www.digium.com/
[asterisk-users] Friday April 3rd Gizmo, OpenSky, Skype for Asterisk, SIP for Skype - where are they?
Hi All,
At the usual time, 12 Noon ET on Friday April 3rd, we expect Michael
Robertson to join the discussion to filed questions about OpenSky and
Gizmo5. I have been testing all of these Skype to X methods except SIP
for Skype since I have no word from them. I can tell you that we've
had good results with bith Skype for Asterisk and OpenSky.
In fact, I am currently accepting calls to my hosted pbx from Skype
and Google Voice via the Gizmo and OpenSky platform and I'm very
pleased with the results. In fact, we may cut our low traffic tollfree
numbers entirely in favor of such services. While not all of these are
free, they are for the most part reasonably-priced. I'll let Michael
discuss this with you. He was recently lambasted on Om Malik's blog
for calling SIP for Skype vaporware. I didn't see his comment as
meaning that, but we'll find out more tomorrow.
Also it's time to give away the Polycom ip450 from e4strategies so we
will be doing that on tomorrow's call as well.
You will need to be on IRC and on the call to be eligible and be
registered with Talkshoe in order to have a PIN:
IRC: #voip-users-conference on Freenode.net or via the web: http://tr.im/vucirc
SIP: 7463#22622#${your_pin_he...@proxy.ideasip.com
SIP g722: You can see this SIP URI generously lent to us by ZipDX in
the title on the IRC channel
PSTN: (724) 444-7444 enter 22622# YOUR_PIN#
See you there!
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To UNSUBSCRIBE or update options visit:
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Re: [asterisk-users] Extract a MOS value from Asterisk CDR
Hello all, I have put my MOS.ods file into
http://dev.leurent.eu/voip/MOS/
My problem is to add the jitter value into the formula
Have you got any idea how to do it?
-- --
Marc LEURENT
Le Thursday 02 April 2009 11.20:06 Mindaugas Kezys, vous avez écrit :
Could you share with us your Openoffice callc function?
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
VoIP Billing and Routing Solutions
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Marc Leurent
Sent: 2009 m. balandžio 2 d. 11:29
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Extract a MOS value from Asterisk CDR
Indeed, we already have
- the function to convert R factor to MOS
- the R function R = R0 -Is-Id-Ie+A
- the codec used
- the rtt, rx/tx jitter, packet loss
What ye do not have but is needed:
- A factor, a note between 0 and 20 - 0 for landlines
- the Burst Ratio, I'm using 1 (random repartition)
I already have an openoffice calc function to calculate the MOS regarding the
rtt, packet loss, codec, I have to add the jitter!
Here are the URL I have used
* http://www.itu.int/rec/T-REC-G.107-200503-S/en
* http://www.ixiacom.com/library/white_papers/display?skey=voip_quality
* http://www.itu.int/ITU-T/studygroups/com12/emodelv1/tut.htm
Have a nice day!
-- --
Marc LEURENT
Ingénieur VoIP
DECKPOINT SA
Une société du groupe VTX Telecom
Rue Eugène-Marziano 15 - 1227 Les Acacias
http://www.vtx.ch - marc.leur...@vtx-telecom.ch
VTX, votre partenaire telecom proche de vous !
Le Thursday 02 April 2009 03.27:48 John Todd, vous avez écrit :
Thank you for the interesting links on MOS values and calculations!
It seems that many (most?) of the values that are used to construct R
and MOS could be obtained from the data that exists within the
dialplan, at least as far as the visible RTP path is concerned. Or
is there data missing in the current RTCP statistics that would be
required to make correct R/MOS value estimates? (If so, then that's
on-topic for asterisk-dev, otherwise this should be moved to asterisk-
users...)
Here is the data that I think is already visible:
- codec choices
- round-trip delay to RTP endpoint
- packet loss
- jitter
I think it is too complex to determine Irecency, A or packet loss
bursts unless there is significant additional code added to Asterisk
to capture more granular time-slices of data on each call. I also
think that mid-call codec changes should not be considered due to
complexity. Currently, I think this is un-necessary since most people
don't even seem to compute MOS to start with.
So in your examination you may come up with a script or dialplan that
creates a synthetic R or MOS value - could you post it to a blog, or
if it is very short, to the asterisk-users mailing list? I think this
would be worthwhile.
JT
On Apr 1, 2009, at 2:57 PM, Mindaugas Kezys wrote:
Sorry for replying for the second time, but this issue is
interesting for me
also.
I found such link: http://www.nessoft.com/kb/50
And this:
http://www.jdsu.com/product-literature/voipstats_an_acc_tm_ae.pdf
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
VoIP Billing and Routing Solutions
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Marc
Leurent
Sent: 2009 m. balandžio 1 d. 18:15
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Extract a MOS value from Asterisk CDR
Hello all,
I'm tring to retrieve a formula to calculate a MOS value from
Asterisk RTCP
stats...
Have you got any idea how to do it?
Thanks
I'm reading all G.107 ITU docs to retrieve something...
I'm saving the SIP RTCP stats with:
[macro-hangupcall]
exten = s,1,Set(CDR(userfield)=${CHANNEL(rtpqos|audio|all)})
exten = s,n,ResetCDR(vw)
exten = s,n,NoCDR()
So I retrieve these values in my MySQL CDR table in order to
calculate a MOS
value:
ssrc
=
592614191;themssrc=0;lp=1;rxjitter=0.00;rxcount=0;txjitter=0.0
0;txcount=20734;rlp=0;rtt=0.094000
codec used: g711a
--
-- --
Marc LEURENT
lf...@leurent.eu
___
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
-- Bandwidth and Colocation Provided by
[asterisk-users] Xorcom and Doorbell
Hi, I am trying to connect a doorbell to a Xorcom device. And the setup is quite simple. But when I push the doorbell all I see on the asterisk cli is: -- Starting simple switch on 'Zap/11-1' [Apr 2 13:00:40] DEBUG[8771]: chan_dahdi.c:6180 ss_thread: not enough digits (and no ambiguous match)... -- Hungup 'Zap/11-1' I defined the extension s,h,i,t,T etc... in my context. Any idea what I might do wrong? Best regards, Loïc. ___ -- 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] evaluate SIP response codes in dialplan
Hi, sorry for joining the discussion so lately. I'd like to ask you to check http://bugs.digium.com/view.php?id=14810. The patch tries to address the issue using channel-variables to propagate the hangup-cause to the calling channel. Best regards, Marcus On Fri, Jan 23, 2009 at 3:08 PM, Johansson Olle E o...@edvina.net wrote: 21 jan 2009 kl. 11.49 skrev Klaus Darilion: Hi Olle! Currently we have the problem that due to SIP-hangupcause-SIP-hangupcause conversions the original hangupcause gets lost in a chain of Asterisk servers using SIP. In chan_sip there is already code for adding the X-Asterisk-Hangupcode header. What about reading this header on the receiving side for setting the hangupcause instead of doing SIP-hangupcause mapping ? In this case we could do that, but there has to be an option to enable it since it will change the behaviour in existing networks. Good idea! /O ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Dipl.-Inf. (FH) Marcus Hunger - hun...@sipgate.de Telefon: +49 (0)211-63 55 55-61 Telefax: +49 (0)211-63 55 55-22 sipgate GmbH - Gladbacher Str. 74 - 40219 Düsseldorf HRB Düsseldorf 39841 - Geschäftsführer: Thilo Salmon, Tim Mois Steuernummer: 106 / 5724 / 7147, Umsatzsteuer-ID: DE219349391 www.sipgate.de - www.sipgate.at - www.sipgate.co.uk ___ -- 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] Extract a MOS value from Asterisk CDR
Formula here: http://www.nessoft.com/kb/50 has jitter in it.
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
VoIP Billing and Routing Solutions
-Original Message-
From: Marc Leurent [mailto:lf...@leurent.eu]
Sent: 2009 m. balandžio 2 d. 13:56
To: asterisk-users@lists.digium.com
Cc: Mindaugas Kezys
Subject: Re: [asterisk-users] Extract a MOS value from Asterisk CDR
Hello all, I have put my MOS.ods file into
http://dev.leurent.eu/voip/MOS/
My problem is to add the jitter value into the formula
Have you got any idea how to do it?
-- --
Marc LEURENT
Le Thursday 02 April 2009 11.20:06 Mindaugas Kezys, vous avez écrit :
Could you share with us your Openoffice callc function?
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
VoIP Billing and Routing Solutions
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Marc Leurent
Sent: 2009 m. balandžio 2 d. 11:29
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Extract a MOS value from Asterisk CDR
Indeed, we already have
- the function to convert R factor to MOS
- the R function R = R0 -Is-Id-Ie+A
- the codec used
- the rtt, rx/tx jitter, packet loss
What ye do not have but is needed:
- A factor, a note between 0 and 20 - 0 for landlines
- the Burst Ratio, I'm using 1 (random repartition)
I already have an openoffice calc function to calculate the MOS regarding the
rtt, packet loss, codec, I have to add the jitter!
Here are the URL I have used
* http://www.itu.int/rec/T-REC-G.107-200503-S/en
* http://www.ixiacom.com/library/white_papers/display?skey=voip_quality
* http://www.itu.int/ITU-T/studygroups/com12/emodelv1/tut.htm
Have a nice day!
-- --
Marc LEURENT
Ingénieur VoIP
DECKPOINT SA
Une société du groupe VTX Telecom
Rue Eugène-Marziano 15 - 1227 Les Acacias
http://www.vtx.ch - marc.leur...@vtx-telecom.ch
VTX, votre partenaire telecom proche de vous !
Le Thursday 02 April 2009 03.27:48 John Todd, vous avez écrit :
Thank you for the interesting links on MOS values and calculations!
It seems that many (most?) of the values that are used to construct R
and MOS could be obtained from the data that exists within the
dialplan, at least as far as the visible RTP path is concerned. Or
is there data missing in the current RTCP statistics that would be
required to make correct R/MOS value estimates? (If so, then that's
on-topic for asterisk-dev, otherwise this should be moved to asterisk-
users...)
Here is the data that I think is already visible:
- codec choices
- round-trip delay to RTP endpoint
- packet loss
- jitter
I think it is too complex to determine Irecency, A or packet loss
bursts unless there is significant additional code added to Asterisk
to capture more granular time-slices of data on each call. I also
think that mid-call codec changes should not be considered due to
complexity. Currently, I think this is un-necessary since most people
don't even seem to compute MOS to start with.
So in your examination you may come up with a script or dialplan that
creates a synthetic R or MOS value - could you post it to a blog, or
if it is very short, to the asterisk-users mailing list? I think this
would be worthwhile.
JT
On Apr 1, 2009, at 2:57 PM, Mindaugas Kezys wrote:
Sorry for replying for the second time, but this issue is
interesting for me
also.
I found such link: http://www.nessoft.com/kb/50
And this:
http://www.jdsu.com/product-literature/voipstats_an_acc_tm_ae.pdf
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
VoIP Billing and Routing Solutions
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Marc
Leurent
Sent: 2009 m. balandžio 1 d. 18:15
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Extract a MOS value from Asterisk CDR
Hello all,
I'm tring to retrieve a formula to calculate a MOS value from
Asterisk RTCP
stats...
Have you got any idea how to do it?
Thanks
I'm reading all G.107 ITU docs to retrieve something...
I'm saving the SIP RTCP stats with:
[macro-hangupcall]
exten = s,1,Set(CDR(userfield)=${CHANNEL(rtpqos|audio|all)})
exten = s,n,ResetCDR(vw)
exten = s,n,NoCDR()
So I retrieve these values in my MySQL CDR table in order to
calculate a MOS
value:
ssrc
=
592614191;themssrc=0;lp=1;rxjitter=0.00;rxcount=0;txjitter=0.0
0;txcount=20734;rlp=0;rtt=0.094000
codec used: g711a
--
-- --
Marc LEURENT
lf...@leurent.eu
[asterisk-users] Mountain ahead of me!
Dear All, Thanks for taking the time to read this. I have been presented with a massive task. I'm not an asterisk expert, but I do know my way around a linux server and infrastructure, and I know when things are not done correctly. A large number of minutes are routed every month, (1m+) and I wish to do this in the most efficient way possible. I've been presented with three linux servers, all in varying states of upkeep, and I've decided, instead of attempting to clean the systems I'm presented with, it is better for me to build a stable platform for asterisk to be migrated onto. This makes my question two fold. 1 What steps should I take, or consider, if I wish to migrate an existing asterisk installation, without it being offline for too long 2 What steps should I look out for, if I wish to move to a MySQL backed for the configuration files, so that I can remove the systems dependence on local configuration. My long term plan is to introduce MySQL to be the backend for the configuration and call log data and put this machine behind a load balancer, so that in due course, when I need to add more machines to handle the load, I will have no need to reconfigure asterisk, or build new configurations, and if I keep the base OS install uniform, I should in theory be able to deploy more asterisk boxes very fast behind a load balancer to increase the capacity of my VoIP Farm with minimal work. *VoIP farm is my term, please do not use it in any presentations to the powers that be inside your organisation - If you wish to do so please send £10(ten) via paypal to my email address which is clearly displayed in the email headers!* Also, in theory, it allows for testing of new configuration, without having to change the configuration on multiple machines at the same time. Which is always a good thing. Any help an advice, or questions are most welcome, as I wish to turn this mountain into a mole hill, a very stable, and expandable mole hill! Thank you for your time, Mr Gabriel ___ -- 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] Xorcom and Doorbell
Loic Didelot wrote: Hi, I am trying to connect a doorbell to a Xorcom device. And the setup is quite simple. But when I push the doorbell all I see on the asterisk cli is: -- Starting simple switch on 'Zap/11-1' [Apr 2 13:00:40] DEBUG[8771]: chan_dahdi.c:6180 ss_thread: not enough digits (and no ambiguous match)... -- Hungup 'Zap/11-1' What number do you have your doorbell configured to dial when the button is pushed? Can you post the context that the doorphone's channel is configured to use? I defined the extension s,h,i,t,T etc... in my context. Any idea what I might do wrong? Your doorbell won't be dialling any of those extensions, of that you can be sure. ___ -- 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] Xorcom and Doorbell
On Thu, Apr 02, 2009 at 01:06:04PM +0200, Loic Didelot wrote: Hi, I am trying to connect a doorbell to a Xorcom device. And the setup is quite simple. But when I push the doorbell all I see on the asterisk cli is: -- Starting simple switch on 'Zap/11-1' [Apr 2 13:00:40] DEBUG[8771]: chan_dahdi.c:6180 ss_thread: not enough digits (and no ambiguous match)... -- Hungup 'Zap/11-1' I defined the extension s,h,i,t,T etc... in my context. Any idea what I might do wrong? Make that extension immediate? (This is what dahdi_genconf generates for it) -- 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
Re: [asterisk-users] 400 calls at g711 how much cpu power
On Thu, Apr 02, 2009 at 09:33:49AM +0100, Gordon Henderson wrote: As for the server - get *everything* in RAM. At least with no disk IO, This is true with respect to e.g. recordings. But most other operations won't bother the disk much. If you have 400 channels doing roughly the same things, the files that they use will mostly be cached. Disabling atime updates (e.g.: noatime, relatime) can help reducing the load of unnecessary writes to the disk. -- 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
Re: [asterisk-users] [Zaptel] Why no driver for PCI voice modems?
Martin wrote: I wonder why people don't get it ? X100P is a winmodem was and always will be. What makes you think anyone doesn't understand that? The problem is the chip on the X100P isn't made any more, and X100P cards are no longer so plentiful. You'll notice the price is going up. They aren't $5 any more. Several Winmodem chips are still readily available, and so are cards containing them. What is missing is someone putting the effort into making drivers for them. 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] [Zaptel] Why no driver for PCI voice modems?
On Thu, Apr 02, 2009 at 08:28:44PM +0800, Steve Underwood wrote: Several Winmodem chips are still readily available, and so are cards containing them. What is missing is someone putting the effort into making drivers for them. Can you list, off the top of your head, modems for which the relevant information is probably available? -- 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
Re: [asterisk-users] Asterisk doesn't relay remote MOH during hold
Furthermore, the following two IETF documents address the need to both signal the hold and provide the music: 1. RFC 5359 (Session Initiation Protocol Service Examples) 2. draft-worley-service-example-03 (Session Initiation Protocol Service Example -- Music on Hold) Unfortunately they both address more complex scenarios and solutions, but they do back me up on the fact that there are good reasons to both signal hold and provide music. R. On Wed, Apr 1, 2009 at 6:16 PM, Richard Brady rnbr...@gmail.com wrote: Hi Tony I can see where you guys are coming from on this and have already enumerated your argument in my own email. But there are very real reasons for a PBX to signal the hold even when it wants to send its own MOH: 1. Bandwidth: under your scheme the PBX would continue to receive bandwidth-consuming media without using it. 2. Privacy: the far-end has an expectation of privacy while on hold and should have the option to mute automatically when held. 3. Feature richness: signalling the hold enables such innovative features such as reverse hold. 4. ISDN interworking: ISDN supports this and SIP should be compatible with that (as per standard ITU-T Q.1912.5) Also, can you explain why the PBX would use a=sendonly but not dispatch media. Why not a=inactive for that case? IMHO, PBX-A would be broken if it passed this along the Hold message to downstream and then started servicing the MOH itself Remember it is not a hold message, it is a media attribute and we are discussing how that should be interpreted within the context of the hold feature in traditional telephony. I would also like to point out in my defence that there are several telephone systems in the field which behave as I described (Nortel BCM50, Aastra Intelligate, Mitel 3300 to name a few). Regards, Richard I have to agree with Kevin on this one. I fail to understand how you have a PBX-A talking to Asterisk talking to PBX-B and the PBX-A placing the call on hold. Typically you should have a Client/Phone to PBX-A to Asterisk to PBX-B to Client/Phone/VoiceMail. If the Client signals Hold, the PBX should NOT be passing that Hold status on but transition audio stream from Client to MOH (assuming MOH is handled). Asterisk shouldn't notice a thing except more RTP packets (or less if it is my teenage daughter on the phone as the case may be). IMHO, PBX-A would be broken if it passed this along the Hold message to downstream and then started servicing the MOH itself on the RTP stream. That just doesn't make sense. Now if PBX-A were not a PBX and were a SIP Router, and the SIP Router was attempting this, I can see how it would Re-Invite, but it shouldn't pass the hold status onto Asterisk. Need some clarity here. Tony Plack ___ -- 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] [Zaptel] Why no driver for PCI voice modems?
Tzafrir Cohen wrote: On Thu, Apr 02, 2009 at 08:28:44PM +0800, Steve Underwood wrote: Several Winmodem chips are still readily available, and so are cards containing them. What is missing is someone putting the effort into making drivers for them. Can you list, off the top of your head, modems for which the relevant information is probably available? Go to the Linmodems mailing list, and look at the things people are getting to work with their Linux machines today. Most of those chips can be programmed for 8k sampling, although as modems they usually sample at 9.6k second (It can simply the maths a bit, even though it means working with more samples). From a quick scan I did a year or two ago, the source code for most of the drivers gives you the starting point you need. 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] async agi question
Async AGI was never released for Asterisk 1.4.X, so probably the patch
you used has a bug or something, do you still have the patch around?
Moy
On Thu, Apr 2, 2009 at 5:44 AM, cyr2...@gmail.com wrote:
Hi Henrik,
I would like to do the same thing you are doing here. I want to implement an
external queue functionality so I need to stop a play file launched
previously with an async agi command on caller's channel, sending the call to
agent's extension.
I'm redirecting caller's channel with REDIRECT while playing is taking place
but I'm always getting a hang up on caller's channel.
I'm using:
asterisk-1.4.18
asterisk-addons-1.4.7
async agi patch 2007-12-11 10:34:12 (the one back-ported to 1.4)
Both caller and agent are using 501 and 500 extensions and the async agi loop
is waiting on 800, for example. The caller is dialing 800 where a play file
is commanded through and async agi stream file command by the application.
The relevant part of extensions.conf follows:
exten = _5.,1,Noop(SIP call on 'sip_sercom' a ${EXTEN});
exten = _5.,n,Wait(1);
exten = _5.,n,Dial(SIP/${EXTEN},${TIMEOUTDIAL},Tto);
exten = _5.,n,Hangup();
exten = _8.,1,Noop(every thing starting 8 ${EXTEN});
exten = _8.,n,AGI(agi:async);
exten = _8.,n,Hangup();
And the redirect command the application is sending to is:
Action: Redirect
Channel: SIP/501-081f0730
Exten: 500
Context: sip_sercom
Priority: 1
Therefore, Henrik, could you show me your related dial plan and the redirect
command you are sending? I wasn't able to see what I'm getting wrong.
thanks in advanced
Jose M Arias
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Re: [asterisk-users] Xorcom and Doorbell
Tadaaah, thanks. immediate=yes fixed it. Loic On Thu, 2009-04-02 at 15:11 +0300, Tzafrir Cohen wrote: On Thu, Apr 02, 2009 at 01:06:04PM +0200, Loic Didelot wrote: Hi, I am trying to connect a doorbell to a Xorcom device. And the setup is quite simple. But when I push the doorbell all I see on the asterisk cli is: -- Starting simple switch on 'Zap/11-1' [Apr 2 13:00:40] DEBUG[8771]: chan_dahdi.c:6180 ss_thread: not enough digits (and no ambiguous match)... -- Hungup 'Zap/11-1' I defined the extension s,h,i,t,T etc... in my context. Any idea what I might do wrong? Make that extension immediate? (This is what dahdi_genconf generates for it) -- Loïc DIDELOT MIXvoip S.a. Tel: +352 20 20 Fax: +352 20 90 ldide...@mixvoip.com http://www.mixvoip.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] activate telco redirection service from Asterisk
Hi, I want my telco to redirect all the incoming calls to my Asterisk towards another number (connected to my old Panasonic PBX) so I can stop Asterisk and repair my office. I tried to send the code *#21# ( Dial(mISDN/1/*#21#) ) but I get a busy channel while it is working with an ISDN phone. How can I do this with Asterisk? Thank you! Giorgio. ___ -- 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] activate telco redirection service from Asterisk
What about *#72#? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Giorgio Incantalupo Sent: Thursday, April 02, 2009 9:01 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] activate telco redirection service from Asterisk Hi, I want my telco to redirect all the incoming calls to my Asterisk towards another number (connected to my old Panasonic PBX) so I can stop Asterisk and repair my office. I tried to send the code *#21# ( Dial(mISDN/1/*#21#) ) but I get a busy channel while it is working with an ISDN phone. How can I do this with Asterisk? Thank you! Giorgio. ___ -- 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] 400 calls at g711 how much cpu power
My only comment is that I am having moral issues with assisting anyone that is planning to call one million phone numbers to play a message and hang up. Doesn't sound like an opt-in kind of campaign to me. When such a thing happens to me on my home phone I get extremely angry. j On Wed, 1 Apr 2009, Erick Perez wrote: We are planning to run an outbound only campaign. A 20-second voice message will be played to callers and our dialer on machine1 will send to machine2-asterisk (1.4) instructions to dial 400 calls, play the message and hang up. This will be done for about 1 million phones. The asterisk box will communicate via SIP to a voice carrier. the voice carrier will then place the calls on pstn. The codec will be g711. So we will never do any transcoding. I have been calculating the CPU power required to do the calls and in previous posting the usual calculation is about 40MHZ per leg when no transcoding is involved. So if we use the 40MHZ rule, we are talking about 40*400=16000MHZ or 1.6Ghz. Comments? -- Erick ___ -- 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] 400 calls at g711 how much cpu power
Yes, we have enough car warranty calls now, just recently joined by the reduce your credit card interest rate calls. :-( Cary -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere Sent: Thursday, April 02, 2009 9:34 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 400 calls at g711 how much cpu power My only comment is that I am having moral issues with assisting anyone that is planning to call one million phone numbers to play a message and hang up. Doesn't sound like an opt-in kind of campaign to me. When such a thing happens to me on my home phone I get extremely angry. j On Wed, 1 Apr 2009, Erick Perez wrote: We are planning to run an outbound only campaign. A 20-second voice message will be played to callers and our dialer on machine1 will send to machine2-asterisk (1.4) instructions to dial 400 calls, play the message and hang up. This will be done for about 1 million phones. The asterisk box will communicate via SIP to a voice carrier. the voice carrier will then place the calls on pstn. The codec will be g711. So we will never do any transcoding. I have been calculating the CPU power required to do the calls and in previous posting the usual calculation is about 40MHZ per leg when no transcoding is involved. So if we use the 40MHZ rule, we are talking about 40*400=16000MHZ or 1.6Ghz. Comments? -- Erick ___ -- 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] Xorcom and Doorbell
Dears How can I send or force sending 180 Ringing instead of 183 back to the caller ?or send both sequential if its impossible I used progressinband=never but it did work . Regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ -- 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] Xorcom and Doorbell
Custom SIP header? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Khaled W. Chehab Sent: Thursday, April 02, 2009 10:02 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Xorcom and Doorbell Dears How can I send or force sending 180 Ringing instead of 183 back to the caller ?or send both sequential if its impossible I used progressinband=never but it did work . Regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ -- 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] Asterisk SIP trunk to Cisco IAD2400
Hi All, Does anyone have a config example for setting up SIP trunking to a CIsco IAD2400 and are willing to share? I've done SIP trunking to Cisco 2600's with PRI's but not to the POTS lines on the IAD's, I'm wondering if that is possible and how to specify the DID on the POTS line config for the IAD. Thanks. JR -- JR Richardson Engineering for the Masses ___ -- 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] SIP 183 progessl
Can you please tell me how to Custom SIP header Regards -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Thursday, April 02, 2009 6:16 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Xorcom and Doorbell Custom SIP header? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Khaled W. Chehab Sent: Thursday, April 02, 2009 10:02 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Xorcom and Doorbell Dears How can I send or force sending 180 Ringing instead of 183 back to the caller ?or send both sequential if its impossible I used progressinband=never but it did work . Regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ -- 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 * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ -- 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] activate telco redirection service from Asterisk
Hi Danny, it is the code to ask the telco your status about the redirection service...when you dial that number, you hear a voice from telco telling you if the redirection has been activated or not. Giorgio Danny Nicholas wrote: What about *#72#? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Giorgio Incantalupo Sent: Thursday, April 02, 2009 9:01 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] activate telco redirection service from Asterisk Hi, I want my telco to redirect all the incoming calls to my Asterisk towards another number (connected to my old Panasonic PBX) so I can stop Asterisk and repair my office. I tried to send the code *#21# ( Dial(mISDN/1/*#21#) ) but I get a busy channel while it is working with an ISDN phone. How can I do this with Asterisk? Thank you! Giorgio. ___ -- 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 -- Giorgio Incantalupo, mailto:gincantal...@fgasoftware.com vo...@work - The Agile PBX http://www.voiceatwork.eu FGA srl - http://www.fgasoftware.com Tel: 02 997663.14, Fax: 02 91390172 ___ -- 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] 400 calls at g711 how much cpu power
Cary Fitch escribió: Yes, we have enough car warranty calls now, just recently joined by the reduce your credit card interest rate calls. :-( Cary It's unbelievable how people use all this marketing strategies that annoy people far away the limit. Fortunately, nobody here in Colombia is doing such a thing (as least on cell phones, because on landlines I've heard cases of calls about winning a car to con people), I would be very angry to receive a call with this type of ugly advertising. I usually accept to receive only call per month, reminding my pendant cell phone bill, and I have enough with all the SMS garbage (sometimes I get three on a day) that I receive from my cell phone operator. If this type of calls problem keeps growing, we would need to maintain an asterisk at home just to block them. Miguel -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere Sent: Thursday, April 02, 2009 9:34 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 400 calls at g711 how much cpu power My only comment is that I am having moral issues with assisting anyone that is planning to call one million phone numbers to play a message and hang up. Doesn't sound like an opt-in kind of campaign to me. When such a thing happens to me on my home phone I get extremely angry. j On Wed, 1 Apr 2009, Erick Perez wrote: We are planning to run an outbound only campaign. A 20-second voice message will be played to callers and our dialer on machine1 will send to machine2-asterisk (1.4) instructions to dial 400 calls, play the message and hang up. This will be done for about 1 million phones. The asterisk box will communicate via SIP to a voice carrier. the voice carrier will then place the calls on pstn. The codec will be g711. So we will never do any transcoding. I have been calculating the CPU power required to do the calls and in previous posting the usual calculation is about 40MHZ per leg when no transcoding is involved. So if we use the 40MHZ rule, we are talking about 40*400=16000MHZ or 1.6Ghz. Comments? -- Erick ___ -- 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 -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center PBX: (+57 1)6500800 ext. 1201 Fax: (+57 1)6500816 Móvil: (+57)3138873587 ___ -- 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] FXO Ignore ring
Is there a way to program an FXO device to totally ignore incoming calls? I want to put an FXO on a Fax line so that 911 calls can be sent via that line, but all other activity on the line is between the Fax machine and the phone company. Perhaps munge the ring tone detect if nothing else? Cary ___ -- 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 183 progessl
Sipaddheader(180 Ringing) might do the trick. If you are compiling your own asterisk, you could change chan_sip.c to replace 183 Session Progress with 180 Ringing (line 3950 in my source) but that might break something else. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Khaled W. Chehab Sent: Thursday, April 02, 2009 10:23 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] SIP 183 progessl Can you please tell me how to Custom SIP header Regards -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Thursday, April 02, 2009 6:16 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Xorcom and Doorbell Custom SIP header? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Khaled W. Chehab Sent: Thursday, April 02, 2009 10:02 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Xorcom and Doorbell Dears How can I send or force sending 180 Ringing instead of 183 back to the caller ?or send both sequential if its impossible I used progressinband=never but it did work . Regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ -- 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 * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ -- 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] FXO Ignore ring
You could use ex-girlfriend logic to hang up the call without answering. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cary Fitch Sent: Thursday, April 02, 2009 10:38 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] FXO Ignore ring Is there a way to program an FXO device to totally ignore incoming calls? I want to put an FXO on a Fax line so that 911 calls can be sent via that line, but all other activity on the line is between the Fax machine and the phone company. Perhaps munge the ring tone detect if nothing else? Cary ___ -- 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] fxotune and the bug
Hi All; I got to know (reading on the wiki) that fxotune was have a bug, and it has been fixed. But I do not know if my current asterisk version contain the fixed one or not? How can I know? My current asterisk version is 1.4.22 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
[asterisk-users] SIP vs RTP destination IP
Is it possible to have asterisk override the connection information embedded in a SIP 200 packet with the registration information? I have multihomed machines with softphones and they register just fine and sip works fine, but the RTP packets get sent to the ip from the SIP connection information and the softphones are sending the wrong ip. I can't find an option in the softphone to change ip it sends. Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 da...@safedatausa.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXO Ignore ring
On Thu, Apr 2, 2009 at 11:37 AM, Cary Fitch ca...@usawide.net wrote: Is there a way to program an FXO device to totally ignore incoming calls? put the port in that context : [incoming-noanswer] exten = s,1,Hangup() hth ___ -- 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] FOR IMMEDIATE RELEASE: NEW CHANNEL DRIVER FOR ASTERISK RELEASED TODAY
Hey ! this can drive to heart attacks randulo a écrit : Nice one, Olle ! :) On Wed, Apr 1, 2009 at 9:18 AM, Olle E. Johansson o...@edvina.net wrote: * NEW CHANNEL DRIVER FOR ASTERISK 1.6 AND VOXSWITCH 3 ADDS AUDIO AND VIDEO TO MICROBLOGGING! ___ -- 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] FXO Ignore ring
- Cary Fitch ca...@usawide.net wrote: Is there a way to program an FXO device to totally ignore incoming calls? I want to put an FXO on a Fax line so that 911 calls can be sent via that line, but all other activity on the line is between the Fax machine and the phone company. Perhaps munge the ring tone detect if nothing else? Cary Greetings Cary- I had the same situation a while back. Please see my post and the answer from another kind user here: http://lists.digium.com/pipermail/asterisk-users/2009-January/224545.html Tim Nelson Systems/Network Support Rockbochs Inc. (218)727-4332 x105 ___ -- 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] Nothing at /proc/zaptel with new Digium TE201
This is a new installation. Here are the specs of my system: Linux asterisk 2.6.21.5-smp #2 SMP Tue Jun 19 14:58:11 CDT 2007 i686 Intel(R) Xeon(R) CPU E5420 @ 2.50GHz GenuineIntel GNU/Linux 08:08.0 Ethernet controller: Digium, Inc. Unknown device 8000 (rev 11) (ethernet?? first time with a card like that for me) dmesg Zapata Telephony Interface Registered on major 196 Zaptel Version: 1.4.12.1 Zaptel Echo Canceller: MG2 proc dev r...@asterisk:/proc/zaptel# ls r...@asterisk:/proc/zaptel# r...@asterisk:/dev/zap# ls -la * crw-rw 1 root root 196, 254 2009-04-02 01:40 channel crw-rw 1 root root 196, 0 2009-04-02 01:40 ctl crw-rw 1 root root 196, 255 2009-04-02 01:40 pseudo crw-rw 1 root root 196, 253 2009-04-02 01:40 timer and genzapconf -l does nothing. Help! I really don't know what is happening. The card is a PCIe And that's everything... I think. ___ -- 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 + Cisco Call Manager
Hi, In our office, we're migrating from a Cisco set up to Asterisk. We'd like to do it gradually, so I've added an asterisk server as an H.323 gateway to the call manager so out going calls are going through asterisk. So far so good. Am now faced with the challenge relaying incoming calls from asterisk to call manager. Has anyone done that before? I won't be allowed to just make the cisco IP phones register with asterisk before it's tested thoroughly and for the gateways to be completely idle, i need to route incoming calls through asterisk. Any hints on how i can achieve this (send calls to cisco call manager 4.1 from an asterisk PBX)? Thanks in advance. Timothy ___ -- 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] Please Advice SIP 183 progessl
Dears
Kindly find my dial script below,I am trying to send the caller 180 ringing
but all tries were failed,
The caller always receive 183 session Progress
Even I add in the sip.conf
progressinband=never
exten = _X.,1,Wait(1)
exten = _X.,n,SetMusicOnHold(English)
exten = _X.,n,WaitMusicOnHold(2)
exten = _X.,n,NoOp(Return-)
exten = _X.,n,Dial(SIP/Gateway1/8430${EXTEN}|300|m)
;exten = _X.,n,SIPAddHeader(Alert-Info: Ring Answer)
exten = _X.,n,Goto(y-${DIALSTATUS},1) ; Jump based on status
(NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten = y-NOANSWER,1,SetMusicOnHold(busy)
exten = y-NOANSWER,n,WaitMusicOnHold(18) ; If busy, Playback Busy /
NOANSWER announce
exten = y-BUSY,1,SetMusicOnHold(busy)
exten = y-BUSY,n,WaitMusicOnHold(18) ; If busy, Playback Busy / busy
announce
exten = _y-.,1,Goto(y-NOANSWER,1) ; Treat anything else as no answer
exten = _X.,n,HangUp()
Please Advice
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Thursday, April 02, 2009 6:40 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] SIP 183 progessl
Sipaddheader(180 Ringing) might do the trick.
If you are compiling your own asterisk, you could change chan_sip.c to
replace 183 Session Progress with 180 Ringing (line 3950 in my source)
but that might break something else.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Khaled W.
Chehab
Sent: Thursday, April 02, 2009 10:23 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] SIP 183 progessl
Can you please tell me how to Custom SIP header
Regards
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Thursday, April 02, 2009 6:16 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Xorcom and Doorbell
Custom SIP header?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Khaled W.
Chehab
Sent: Thursday, April 02, 2009 10:02 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Xorcom and Doorbell
Dears
How can I send or force sending 180 Ringing instead of 183 back to the
caller ?or send both sequential if its impossible
I used progressinband=never but it did work .
Regards
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No employee or agent is authorized to conclude any binding agreement on
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attachments, kindly delete it immediately from your system and notify the
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Xplorium does not guarantee the integrity of this
Re: [asterisk-users] Nothing at /proc/zaptel with new Digium TE201
criptos wrote: This is a new installation. 08:08.0 Ethernet controller: Digium, Inc. Unknown device 8000 (rev 11) (ethernet?? first time with a card like that for me) dmesg Zapata Telephony Interface Registered on major 196 Zaptel Version: 1.4.12.1 Zaptel Echo Canceller: MG2 proc dev r...@asterisk:/proc/zaptel# ls r...@asterisk:/proc/zaptel# r...@asterisk:/dev/zap# ls -la * crw-rw 1 root root 196, 254 2009-04-02 01:40 channel crw-rw 1 root root 196, 0 2009-04-02 01:40 ctl crw-rw 1 root root 196, 255 2009-04-02 01:40 pseudo crw-rw 1 root root 196, 253 2009-04-02 01:40 timer Help! I really don't know what is happening. The card is a PCIe It appears as if you have a TE121 installed in your system which is serviced by the wcte12xp driver. Does 'lsmod | grep wcte12xp' show that the wcte12xp driver is loaded? If not, what happens when you run 'modprobe wcte12xp'? -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] fxotune and the bug
bilal ghayyad wrote: Hi All; I got to know (reading on the wiki) that fxotune was have a bug, and it has been fixed. But I do not know if my current asterisk version contain the fixed one or not? How can I know? My current asterisk version is 1.4.22 Current version of fxotune (in current 1.4 Zaptel and DAHDI) does not have any outstanding bugs. From a quick glance over the wiki page, it looks like it has some interesting information, but a lot of it is out of date. My guess is the bug you're referring to is the one that says it has problems with dialtone detection or something of that nature. The most current version of fxotune is pretty much immune to dialtone or other background noise due to the newer way it does signal measurement (using frequency analysis instead of frequency agnostic power calculation), so you shouldn't see any problems with this. Matthew Fredrickson Digium, Inc. ___ -- 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] Magic! Re: Nothing at /proc/zaptel with new Digium TE201
Holy crap! This list is magical :D it started working... maybe yesterday was too late when I was configuring this thing On Thursday 02 April 2009 09:54:18 criptos wrote: This is a new installation. Here are the specs of my system: Linux asterisk 2.6.21.5-smp #2 SMP Tue Jun 19 14:58:11 CDT 2007 i686 Intel(R) Xeon(R) CPU E5420 @ 2.50GHz GenuineIntel GNU/Linux 08:08.0 Ethernet controller: Digium, Inc. Unknown device 8000 (rev 11) (ethernet?? first time with a card like that for me) dmesg Zapata Telephony Interface Registered on major 196 Zaptel Version: 1.4.12.1 Zaptel Echo Canceller: MG2 proc dev r...@asterisk:/proc/zaptel# ls r...@asterisk:/proc/zaptel# r...@asterisk:/dev/zap# ls -la * crw-rw 1 root root 196, 254 2009-04-02 01:40 channel crw-rw 1 root root 196, 0 2009-04-02 01:40 ctl crw-rw 1 root root 196, 255 2009-04-02 01:40 pseudo crw-rw 1 root root 196, 253 2009-04-02 01:40 timer and genzapconf -l does nothing. Help! I really don't know what is happening. The card is a PCIe And that's everything... I think. ___ -- 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] opermode=?
Hi All; If I need to set the opermode to King Saudi Arabia, what the name I have to use? For example, to set it for kuwait then I use opermode=KUWAIT. So what will be for Saudi Arabia? 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] Magic! Re: Nothing at /proc/zaptel with new DigiumTE201
Did you reboot? Zaptel does not work until you reboot or do a manual modprobe. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of criptos Sent: Thursday, April 02, 2009 11:18 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Magic! Re: Nothing at /proc/zaptel with new DigiumTE201 Holy crap! This list is magical :D it started working... maybe yesterday was too late when I was configuring this thing On Thursday 02 April 2009 09:54:18 criptos wrote: This is a new installation. Here are the specs of my system: Linux asterisk 2.6.21.5-smp #2 SMP Tue Jun 19 14:58:11 CDT 2007 i686 Intel(R) Xeon(R) CPU E5420 @ 2.50GHz GenuineIntel GNU/Linux 08:08.0 Ethernet controller: Digium, Inc. Unknown device 8000 (rev 11) (ethernet?? first time with a card like that for me) dmesg Zapata Telephony Interface Registered on major 196 Zaptel Version: 1.4.12.1 Zaptel Echo Canceller: MG2 proc dev r...@asterisk:/proc/zaptel# ls r...@asterisk:/proc/zaptel# r...@asterisk:/dev/zap# ls -la * crw-rw 1 root root 196, 254 2009-04-02 01:40 channel crw-rw 1 root root 196, 0 2009-04-02 01:40 ctl crw-rw 1 root root 196, 255 2009-04-02 01:40 pseudo crw-rw 1 root root 196, 253 2009-04-02 01:40 timer and genzapconf -l does nothing. Help! I really don't know what is happening. The card is a PCIe And that's everything... I think. ___ -- 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] fxotune and the bug
Dear Mathew; Kindly find the link of the batch tha fixed the bug: http://bugs.digium.com/view.php?id=7136 It is written that last update was in 2008-06-07 11:36, so for that I do not know if my asterisk and zaptel versions include this fix or not? Because I installed them before this date. How can I know starting from which version this patch has been included? Any advise. Regards Bilal --- On Thu, 4/2/09, Matthew Fredrickson cres...@digium.com wrote: From: Matthew Fredrickson cres...@digium.com Subject: Re: [asterisk-users] fxotune and the bug To: bilmar...@yahoo.com, Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Thursday, April 2, 2009, 12:17 PM bilal ghayyad wrote: Hi All; I got to know (reading on the wiki) that fxotune was have a bug, and it has been fixed. But I do not know if my current asterisk version contain the fixed one or not? How can I know? My current asterisk version is 1.4.22 Current version of fxotune (in current 1.4 Zaptel and DAHDI) does not have any outstanding bugs. From a quick glance over the wiki page, it looks like it has some interesting information, but a lot of it is out of date. My guess is the bug you're referring to is the one that says it has problems with dialtone detection or something of that nature. The most current version of fxotune is pretty much immune to dialtone or other background noise due to the newer way it does signal measurement (using frequency analysis instead of frequency agnostic power calculation), so you shouldn't see any problems with this. Matthew Fredrickson Digium, Inc. ___ -- 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] fxotune and the bug
bilal ghayyad wrote: Dear Mathew; Kindly find the link of the batch tha fixed the bug: http://bugs.digium.com/view.php?id=7136 It is written that last update was in 2008-06-07 11:36, so for that I do not know if my asterisk and zaptel versions include this fix or not? Because I installed them before this date. How can I know starting from which version this patch has been included? That particular patch is old and out of date and does not have the latest fixes that include the background noise and tone immunity code. If your problem is that you simply don't want to update Zaptel though, you can build use the fxotune utility from the latest version of Zaptel and just don't run make install so you don't overwrite your existing Zaptel. Matthew Fredrickson Digium, Inc. Any advise. Regards Bilal --- On Thu, 4/2/09, Matthew Fredrickson cres...@digium.com wrote: From: Matthew Fredrickson cres...@digium.com Subject: Re: [asterisk-users] fxotune and the bug To: bilmar...@yahoo.com, Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Thursday, April 2, 2009, 12:17 PM bilal ghayyad wrote: Hi All; I got to know (reading on the wiki) that fxotune was have a bug, and it has been fixed. But I do not know if my current asterisk version contain the fixed one or not? How can I know? My current asterisk version is 1.4.22 Current version of fxotune (in current 1.4 Zaptel and DAHDI) does not have any outstanding bugs. From a quick glance over the wiki page, it looks like it has some interesting information, but a lot of it is out of date. My guess is the bug you're referring to is the one that says it has problems with dialtone detection or something of that nature. The most current version of fxotune is pretty much immune to dialtone or other background noise due to the newer way it does signal measurement (using frequency analysis instead of frequency agnostic power calculation), so you shouldn't see any problems with this. Matthew Fredrickson Digium, Inc. ___ -- 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 183 progessl
On Thu, 2009-04-02 at 10:40 -0500, Danny Nicholas wrote: Sipaddheader(180 Ringing) might do the trick. Danny, I appreciate your enthusiasm for helping people on the mailing list, but unfortunately this is not the correct method of doing what the original poster is asking about. It's not enough to add a custom SIP header... what he really wants is a SIP response, not a SIP header. Let me see if I can shed a bit more light on the original question. To send a SIP 183 message (with early media), you can use the Playback applications with the noanswer option. Here's a quick example: exten = 123,1,Playback(pls-hold-while-try,noanswer) exten = 123,n,Dial(SIP/sip_peer,20) If you were to dial this extension from a SIP device, you'd see that you'd first get a 183 with early media, and then you'd later get the 200 OK (assuming that SIP/sip_peer answered the call). -- Jared Smith Training Manager Digium, Inc. ___ -- 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] Magic List: Thanks Shain Rufeel Danny Nicholas.
Thanks Actually, I tried everything before... I even do a genzapconf -d. I really don't know what happened. After I posted to the list, I made another genzapconf -d and then genzapconf - l and the card just appeared. Don't know the exact reason, but... it's working (the card, I still need to migrate the asterisk from one machine to other) ___ -- 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] T1/PRI ignore answer signal
Is there anyway a T1/PRI can ignore the ANSWERED signal and just go straight from a dial command to the call was answered? I have a PBX that when calling a certain analog trunk it is not giving me signaling that the call was answered however I hear the PA system come off hook and give dial tone. So is there something in the dial command that can override the signaling and just go straight to the answered state so my AGI runs? The T1 is functioning normal for all other calls. Its just to this analog trunk that has the PA system connected to it. Jerry ___ -- 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] meetme dahdi and zaptel
We recently updated our Asterisk (1.4.24) box from Zaptel (1.4.12.1) to Dahdi (2.1.0.4). Everything seemed to go smooth with the exception of meetme. Meetme seems to not be able to find a zap channel for conferencing. We use voice introductions in our conference bridge and it seems to break that feature. The error from the console is # app_meetme.c:2593 find_conf: No Zap channel available for conference, user introduction disabled I've added... dahdichanname=no to /etc/asterisk/asterisk.conf My thought was that would allow Asterisk to use Dahdi just as if it was ZAP. asterisk*CLI zap show status Description Alarms IRQbpviol CRC4 T2XXP (PCI) Card 0 Span 1OK 0 0 0 T2XXP (PCI) Card 0 Span 2RED0 0 0 Span 2 is expected to be down as we don't have it connected to anything. asterisk*CLI zap show channels Chan Extension Context Language MOH Interpret 1incoming_pstn default 2incoming_pstn default 3incoming_pstn default 4incoming_pstn default 5incoming_pstn default 6incoming_pstn default 7incoming_pstn default 8incoming_pstn default 9incoming_pstn default 10incoming_pstn default 11incoming_pstn default 12incoming_pstn default 13incoming_pstn default 14incoming_pstn default 15incoming_pstn default 16incoming_pstn default 17incoming_pstn default 18incoming_pstn default 19incoming_pstn default 20incoming_pstn default 21incoming_pstn default 22incoming_pstn default 23incoming_pstn default Should Zaptel be fully removed prior to Asterisk being compiled? It seems that something with the meetme app is still linked somehow to Zaptel. Has anyone else come across this? Any suggestions? Thanks, Dave ___ -- 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 Advice SIP 183 progessl
Guys I tried the R,r options at the DIAL(SIP/,,rm) and also its sending
183
Any Advice
Dears
Kindly find my dial script below,I am trying to send the caller 180 ringing
but all tries were failed,
The caller always receive 183 session Progress
Even I add in the sip.conf
progressinband=never
exten = _X.,1,Wait(1)
exten = _X.,n,SetMusicOnHold(English)
exten = _X.,n,WaitMusicOnHold(2)
exten = _X.,n,NoOp(Return-)
exten = _X.,n,Dial(SIP/Gateway1/8430${EXTEN}|300|m)
;exten = _X.,n,SIPAddHeader(Alert-Info: Ring Answer)
exten = _X.,n,Goto(y-${DIALSTATUS},1) ; Jump based on status
(NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten = y-NOANSWER,1,SetMusicOnHold(busy)
exten = y-NOANSWER,n,WaitMusicOnHold(18) ; If busy, Playback Busy /
NOANSWER announce
exten = y-BUSY,1,SetMusicOnHold(busy)
exten = y-BUSY,n,WaitMusicOnHold(18) ; If busy, Playback Busy / busy
announce
exten = _y-.,1,Goto(y-NOANSWER,1) ; Treat anything else as no answer
exten = _X.,n,HangUp()
Please Advice
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Thursday, April 02, 2009 6:40 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] SIP 183 progessl
Sipaddheader(180 Ringing) might do the trick.
If you are compiling your own asterisk, you could change chan_sip.c to
replace 183 Session Progress with 180 Ringing (line 3950 in my source)
but that might break something else.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Khaled W.
Chehab
Sent: Thursday, April 02, 2009 10:23 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] SIP 183 progessl
Can you please tell me how to Custom SIP header
Regards
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Thursday, April 02, 2009 6:16 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Xorcom and Doorbell
Custom SIP header?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Khaled W.
Chehab
Sent: Thursday, April 02, 2009 10:02 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Xorcom and Doorbell
Dears
How can I send or force sending 180 Ringing instead of 183 back to the
caller ?or send both sequential if its impossible
I used progressinband=never but it did work .
Regards
*
No employee or agent is authorized to conclude any binding agreement on
behalf of Xplorium with another party by e-mail without express written
confirmation by an officer of Xplorium. Any views expressed by an individual
in this electronic message do not necessarily reflect views of Xplorium or
its subsidiaries and associates.
This electronic message and its attachments are solely addressed to the
addressee(s), and contain confidential information protected from disclosure
belonging to Xplorium.
If you are not the intended addressee of this electronic message and its
attachments, kindly delete it immediately from your system and notify the
sender by electronic mail. You must not copy this message or attachment or
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Xplorium does not guarantee the integrity of this electronic message and any
of its attachments, or that they are free from computer viruses or other
defects.
*
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*
No employee or agent is authorized to conclude any binding agreement on
behalf of Xplorium with another party by e-mail without express written
confirmation by an officer of Xplorium. Any views expressed by an individual
in this electronic message do not necessarily reflect views of Xplorium or
its subsidiaries and associates.
This electronic message and its attachments are solely addressed to the
addressee(s), and contain confidential information protected from disclosure
belonging to Xplorium.
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attachments, kindly delete it immediately from your system and notify the
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Re: [asterisk-users] SIP 183 progessl
Thanks for not being too critical and for providing a good clarification. I've been doing Asterisk for about 7 months now and realize that my answer might or might not be correct. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jared Smith Sent: Thursday, April 02, 2009 11:41 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SIP 183 progessl On Thu, 2009-04-02 at 10:40 -0500, Danny Nicholas wrote: Sipaddheader(180 Ringing) might do the trick. Danny, I appreciate your enthusiasm for helping people on the mailing list, but unfortunately this is not the correct method of doing what the original poster is asking about. It's not enough to add a custom SIP header... what he really wants is a SIP response, not a SIP header. Let me see if I can shed a bit more light on the original question. To send a SIP 183 message (with early media), you can use the Playback applications with the noanswer option. Here's a quick example: exten = 123,1,Playback(pls-hold-while-try,noanswer) exten = 123,n,Dial(SIP/sip_peer,20) If you were to dial this extension from a SIP device, you'd see that you'd first get a 183 with early media, and then you'd later get the 200 OK (assuming that SIP/sip_peer answered the call). -- Jared Smith Training Manager Digium, Inc. ___ -- 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 Advice SIP 183 progessl
This is a hack-fix but if you Answer the call before dialing, that might
remove the progress message
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Khaled W.
Chehab
Sent: Thursday, April 02, 2009 12:47 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Please Advice SIP 183 progessl
Guys I tried the R,r options at the DIAL(SIP/,,rm) and also its sending
183
Any Advice
Dears
Kindly find my dial script below,I am trying to send the caller 180 ringing
but all tries were failed,
The caller always receive 183 session Progress
Even I add in the sip.conf
progressinband=never
exten = _X.,1,Wait(1)
exten = _X.,n,SetMusicOnHold(English)
exten = _X.,n,WaitMusicOnHold(2)
exten = _X.,n,NoOp(Return-)
-- add --
Exten = _X.,n,Answer()
-- end add --
exten = _X.,n,Dial(SIP/Gateway1/8430${EXTEN}|300|m)
;exten = _X.,n,SIPAddHeader(Alert-Info: Ring Answer)
exten = _X.,n,Goto(y-${DIALSTATUS},1) ; Jump based on status
(NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten = y-NOANSWER,1,SetMusicOnHold(busy)
exten = y-NOANSWER,n,WaitMusicOnHold(18) ; If busy, Playback Busy /
NOANSWER announce
exten = y-BUSY,1,SetMusicOnHold(busy)
exten = y-BUSY,n,WaitMusicOnHold(18) ; If busy, Playback Busy / busy
announce
exten = _y-.,1,Goto(y-NOANSWER,1) ; Treat anything else as no answer
exten = _X.,n,HangUp()
Please Advice
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Thursday, April 02, 2009 6:40 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] SIP 183 progessl
Sipaddheader(180 Ringing) might do the trick.
If you are compiling your own asterisk, you could change chan_sip.c to
replace 183 Session Progress with 180 Ringing (line 3950 in my source)
but that might break something else.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Khaled W.
Chehab
Sent: Thursday, April 02, 2009 10:23 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] SIP 183 progessl
Can you please tell me how to Custom SIP header
Regards
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Thursday, April 02, 2009 6:16 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Xorcom and Doorbell
Custom SIP header?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Khaled W.
Chehab
Sent: Thursday, April 02, 2009 10:02 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Xorcom and Doorbell
Dears
How can I send or force sending 180 Ringing instead of 183 back to the
caller ?or send both sequential if its impossible
I used progressinband=never but it did work .
Regards
*
No employee or agent is authorized to conclude any binding agreement on
behalf of Xplorium with another party by e-mail without express written
confirmation by an officer of Xplorium. Any views expressed by an individual
in this electronic message do not necessarily reflect views of Xplorium or
its subsidiaries and associates.
This electronic message and its attachments are solely addressed to the
addressee(s), and contain confidential information protected from disclosure
belonging to Xplorium.
If you are not the intended addressee of this electronic message and its
attachments, kindly delete it immediately from your system and notify the
sender by electronic mail. You must not copy this message or attachment or
disclose its content to any other person.
Xplorium does not guarantee the integrity of this electronic message and any
of its attachments, or that they are free from computer viruses or other
defects.
*
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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*
No employee or agent is authorized to conclude any binding agreement on
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confirmation by an officer of Xplorium. Any views expressed by an individual
in this electronic message do not
[asterisk-users] VB6 to HUD Pro Integration
Hello All, Is there anyone out there that is able to integrate a custom visual basic 6 application to Fonality’s Trixbox HUD Pro? Thanks, Greg ___ -- 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 G729 codec...
Humm... should the list would be magic again? I have just intsalled, using the register, benchmark and downloared the correct codec to my asterisk installation, but I don't have the g729 command at my CLI... Any advice... Do I reboot? ;D ___ -- 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 Advice SIP 183 progessl
I tried it but it didn't work even ,If I use Answer() function , Billing
will starts
Thanks
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Thursday, April 02, 2009 8:59 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Please Advice SIP 183 progessl
This is a hack-fix but if you Answer the call before dialing, that might
remove the progress message
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Khaled W.
Chehab
Sent: Thursday, April 02, 2009 12:47 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Please Advice SIP 183 progessl
Guys I tried the R,r options at the DIAL(SIP/,,rm) and also its sending
183
Any Advice
Dears
Kindly find my dial script below,I am trying to send the caller 180 ringing
but all tries were failed,
The caller always receive 183 session Progress
Even I add in the sip.conf
progressinband=never
exten = _X.,1,Wait(1)
exten = _X.,n,SetMusicOnHold(English)
exten = _X.,n,WaitMusicOnHold(2)
exten = _X.,n,NoOp(Return-)
-- add --
Exten = _X.,n,Answer()
-- end add --
exten = _X.,n,Dial(SIP/Gateway1/8430${EXTEN}|300|m)
;exten = _X.,n,SIPAddHeader(Alert-Info: Ring Answer)
exten = _X.,n,Goto(y-${DIALSTATUS},1) ; Jump based on status
(NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten = y-NOANSWER,1,SetMusicOnHold(busy)
exten = y-NOANSWER,n,WaitMusicOnHold(18) ; If busy, Playback Busy /
NOANSWER announce
exten = y-BUSY,1,SetMusicOnHold(busy)
exten = y-BUSY,n,WaitMusicOnHold(18) ; If busy, Playback Busy / busy
announce
exten = _y-.,1,Goto(y-NOANSWER,1) ; Treat anything else as no answer
exten = _X.,n,HangUp()
Please Advice
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Thursday, April 02, 2009 6:40 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] SIP 183 progessl
Sipaddheader(180 Ringing) might do the trick.
If you are compiling your own asterisk, you could change chan_sip.c to
replace 183 Session Progress with 180 Ringing (line 3950 in my source)
but that might break something else.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Khaled W.
Chehab
Sent: Thursday, April 02, 2009 10:23 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] SIP 183 progessl
Can you please tell me how to Custom SIP header
Regards
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Thursday, April 02, 2009 6:16 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Xorcom and Doorbell
Custom SIP header?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Khaled W.
Chehab
Sent: Thursday, April 02, 2009 10:02 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Xorcom and Doorbell
Dears
How can I send or force sending 180 Ringing instead of 183 back to the
caller ?or send both sequential if its impossible
I used progressinband=never but it did work .
Regards
*
No employee or agent is authorized to conclude any binding agreement on
behalf of Xplorium with another party by e-mail without express written
confirmation by an officer of Xplorium. Any views expressed by an individual
in this electronic message do not necessarily reflect views of Xplorium or
its subsidiaries and associates.
This electronic message and its attachments are solely addressed to the
addressee(s), and contain confidential information protected from disclosure
belonging to Xplorium.
If you are not the intended addressee of this electronic message and its
attachments, kindly delete it immediately from your system and notify the
sender by electronic mail. You must not copy this message or attachment or
disclose its content to any other person.
Xplorium does not guarantee the integrity of this electronic message and any
of its attachments, or that they are free from computer viruses or other
defects.
*
___
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
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asterisk-users mailing list
Re: [asterisk-users] Asterisk G729 codec...
You should not have a G729 command on the CLI. Codecs are addressed in sip.conf, dahdi.conf, etc. restarting Asterisk might do the trick. You only need to reboot for a driver level change. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of criptos Sent: Thursday, April 02, 2009 1:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk G729 codec... Humm... should the list would be magic again? I have just intsalled, using the register, benchmark and downloared the correct codec to my asterisk installation, but I don't have the g729 command at my CLI... Any advice... Do I reboot? ;D ___ -- 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] Asterisk G729 codec...
Danny Nicholas wrote: You should not have a G729 command on the CLI. Codecs are addressed in sip.conf, dahdi.conf, etc. restarting Asterisk might do the trick. You only need to reboot for a driver level change. This is incorrect. Digium's codec_g729a.so module does in fact add a 'g729 show' command to the CLI, when it has found at least one valid license file. so that the user can see how many of their licensed channels are in use. If the 'g729 show' command is not available after you have loaded the module, then you need to look closely at your Asterisk log files because the module was not able to find any valid license files. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Please Advice SIP 183 progessl
Try replacing answer() with playback(tt-monkeys)
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Khaled W.
Chehab
Sent: Thursday, April 02, 2009 1:33 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Please Advice SIP 183 progessl
I tried it but it didn't work even ,If I use Answer() function , Billing
will starts
Thanks
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Thursday, April 02, 2009 8:59 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Please Advice SIP 183 progessl
This is a hack-fix but if you Answer the call before dialing, that might
remove the progress message
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Khaled W.
Chehab
Sent: Thursday, April 02, 2009 12:47 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Please Advice SIP 183 progessl
Guys I tried the R,r options at the DIAL(SIP/,,rm) and also its sending
183
Any Advice
Dears
Kindly find my dial script below,I am trying to send the caller 180 ringing
but all tries were failed,
The caller always receive 183 session Progress
Even I add in the sip.conf
progressinband=never
exten = _X.,1,Wait(1)
exten = _X.,n,SetMusicOnHold(English)
exten = _X.,n,WaitMusicOnHold(2)
exten = _X.,n,NoOp(Return-)
-- add --
Exten = _X.,n,Answer()
-- end add --
exten = _X.,n,Dial(SIP/Gateway1/8430${EXTEN}|300|m)
;exten = _X.,n,SIPAddHeader(Alert-Info: Ring Answer)
exten = _X.,n,Goto(y-${DIALSTATUS},1) ; Jump based on status
(NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten = y-NOANSWER,1,SetMusicOnHold(busy)
exten = y-NOANSWER,n,WaitMusicOnHold(18) ; If busy, Playback Busy /
NOANSWER announce
exten = y-BUSY,1,SetMusicOnHold(busy)
exten = y-BUSY,n,WaitMusicOnHold(18) ; If busy, Playback Busy / busy
announce
exten = _y-.,1,Goto(y-NOANSWER,1) ; Treat anything else as no answer
exten = _X.,n,HangUp()
Please Advice
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Thursday, April 02, 2009 6:40 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] SIP 183 progessl
Sipaddheader(180 Ringing) might do the trick.
If you are compiling your own asterisk, you could change chan_sip.c to
replace 183 Session Progress with 180 Ringing (line 3950 in my source)
but that might break something else.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Khaled W.
Chehab
Sent: Thursday, April 02, 2009 10:23 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] SIP 183 progessl
Can you please tell me how to Custom SIP header
Regards
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Thursday, April 02, 2009 6:16 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Xorcom and Doorbell
Custom SIP header?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Khaled W.
Chehab
Sent: Thursday, April 02, 2009 10:02 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Xorcom and Doorbell
Dears
How can I send or force sending 180 Ringing instead of 183 back to the
caller ?or send both sequential if its impossible
I used progressinband=never but it did work .
Regards
*
No employee or agent is authorized to conclude any binding agreement on
behalf of Xplorium with another party by e-mail without express written
confirmation by an officer of Xplorium. Any views expressed by an individual
in this electronic message do not necessarily reflect views of Xplorium or
its subsidiaries and associates.
This electronic message and its attachments are solely addressed to the
addressee(s), and contain confidential information protected from disclosure
belonging to Xplorium.
If you are not the intended addressee of this electronic message and its
attachments, kindly delete it immediately from your system and notify the
sender by electronic mail. You must not copy this message or attachment or
disclose its content to any other person.
Xplorium does not guarantee the integrity of this electronic message and any
of its attachments, or that they are free from computer viruses or other
defects.
*
Re: [asterisk-users] Asterisk G729 codec...
Documenation show that at the asterisk cli you can use the g729 show to show the codec usage/license availability... This is what is missing, So, I'm not sure that my licenses are being loaded. On Thursday 02 April 2009 12:35:18 Danny Nicholas wrote: You should not have a G729 command on the CLI. Codecs are addressed in sip.conf, dahdi.conf, etc. restarting Asterisk might do the trick. You only need to reboot for a driver level change. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of criptos Sent: Thursday, April 02, 2009 1:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk G729 codec... Humm... should the list would be magic again? I have just intsalled, using the register, benchmark and downloared the correct codec to my asterisk installation, but I don't have the g729 command at my CLI... Any advice... Do I reboot? ;D ___ -- 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
[asterisk-users] Asterisk 1.2.32, 1.4.24.1, and 1.6.0.8 Now Available
The Asterisk.org development team has released Asterisk versions 1.2.32, 1.4.24.1, and 1.6.0.8. These releases are available for immediate download from http://downloads.digium.com/. This update for Asterisk includes a security fix for chan_sip. Please see the associated security advisory for more details: http://downloads.digium.com/pub/security/AST-2009-003.html. This security issue affects the 1.2, 1.4 and 1.6.0 versions of Asterisk. Thank you for your continued support of Asterisk! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AST-2009-003: SIP responses expose valid usernames
Asterisk Project Security Advisory - AST-2009-003 ++ | Product | Asterisk | |+---| | Summary | SIP responses expose valid usernames | |+---| | Nature of Advisory | Information leak | |+---| | Susceptibility | Remote Unauthenticated Sessions | |+---| | Severity | Minor | |+---| | Exploits Known | No| |+---| |Reported On | February 23, 2009 | |+---| |Reported By | Gentoo Linux Project: Kerin Millar ( kerframil on | || irc.freenode.net ) and Fergal Glynn FGlynn AT | || veracode DOT com | |+---| | Posted On | April 2, 2009 | |+---| | Last Updated On | April 2, 2009 | |+---| | Advisory Contact | Tilghman Lesher tlesher AT digium DOT com | |+---| | CVE Name | CVE-2008-3903 | ++ ++ | Description | In 2006, the Asterisk maintainers made it more difficult | | | to scan for valid SIP usernames by implementing an | | | option called alwaysauthreject, which should return a | | | 401 error on all replies which are generated for users | | | which do not exist. While this was sufficient at the | | | time, due to ever increasing compliance with RFC 3261, | | | the SIP specification, that is no longer sufficient as a | | | means towards preventing attackers from checking | | | responses to verify whether a SIP account exists on a| | | machine. | | | | | | What we have done is to carefully emulate exactly the| | | same responses throughout possible dialogs, which should | | | prevent attackers from gleaning this information. All| | | invalid users, if this option is turned on, will receive | | | the same response throughout the dialog, as if a | | | username was valid, but the password was incorrect. | | | | | | It is important to note several things. First, this | | | vulnerability is derived directly from the SIP | | | specification, and it is a technical violation of RFC| | | 3261 (and subsequent RFCs, as of this date), for us to | | | return these responses. Second, this attack is made much | | | more difficult if administrators avoided creating| | | all-numeric usernames and especially all-numeric | | | passwords. This combination is extremely vulnerable for | | | servers connected to the public Internet, even with this | | | patch in place. While it may make configuring SIP| | | telephones easier in the short term, it has the | | | potential to cause grief over the long term. | ++ ++ | Resolution | Upgrade to one of the versions below, or apply one of the | || patches specified in the Patches section. | ++
Re: [asterisk-users] Nothing at /proc/zaptel with new Digium TE201
On Thu, Apr 02, 2009 at 09:54:18AM -0600, criptos wrote: This is a new installation. Here are the specs of my system: Linux asterisk 2.6.21.5-smp #2 SMP Tue Jun 19 14:58:11 CDT 2007 i686 Intel(R) Xeon(R) CPU E5420 @ 2.50GHz GenuineIntel GNU/Linux 08:08.0 Ethernet controller: Digium, Inc. Unknown device 8000 (rev 11) (ethernet?? first time with a card like that for me) dmesg Zapata Telephony Interface Registered on major 196 Zaptel Version: 1.4.12.1 Zaptel Echo Canceller: MG2 proc dev r...@asterisk:/proc/zaptel# ls r...@asterisk:/proc/zaptel# Being in that directory is generally not a good idea: A bit like: http://docs.tzafrir.org.il/dahdi-tools/README.Astribank.html#_empty_proc_dir Nothing critical relies on /proc/zaptel . genzaptelconf / zapconf do use that information, and if it has failed to create, they will see nothing. -- 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
Re: [asterisk-users] Asterisk G729 codec...
criptos escribió: Humm... should the list would be magic again? I have just intsalled, using the register, benchmark and downloared the correct codec to my asterisk installation, but I don't have the g729 command at my CLI... Any advice... Do I reboot? ;D Did you load your brand new codec_g729.so module after putting it on /usr/lib/asterisk/modules? CLI module load codec_g729.so What does the command core show codecs show? Cheers, ___ -- 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 -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center PBX: (+57 1)6500800 ext. 1201 Fax: (+57 1)6500816 Móvil: (+57)3138873587 ___ -- 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] 'codec_g729a.so' does not provide a description Re: Asterisk G729 codec...
codec is on place: r...@asterisk:/usr/lib/asterisk/modules# ls chan_* chan_agent.so* chan_iax2.so* chan_mgcp.so* chan_phone.so* chan_skinny.so* chan_zap.so* chan_alsa.so* chan_local.so* chan_oss.so* chan_sip.so* chan_unicall.so* r...@asterisk:~# asterisk -v asterisk.log [1]+ Stopped asterisk -v asterisk.log r...@asterisk:~# bg [1]+ asterisk -v asterisk.log r...@asterisk:~# asterisk -rx 'stop now' Disconnected from Asterisk server [1]+ Doneasterisk -v asterisk.log r...@asterisk:~# r...@asterisk:~# grep 729 asterisk.log [Apr 2 13:57:38] WARNING[3662]: loader.c:605 inspect_module: Module 'codec_g729a.so' does not provide a description. [Apr 2 13:57:38] WARNING[3662]: loader.c:662 load_resource: Module 'codec_g729a.so' could not be loaded. This is the message that I get. On Thursday 02 April 2009 12:42:58 Kevin P. Fleming wrote: Danny Nicholas wrote: You should not have a G729 command on the CLI. Codecs are addressed in sip.conf, dahdi.conf, etc. restarting Asterisk might do the trick. You only need to reboot for a driver level change. This is incorrect. Digium's codec_g729a.so module does in fact add a 'g729 show' command to the CLI, when it has found at least one valid license file. so that the user can see how many of their licensed channels are in use. If the 'g729 show' command is not available after you have loaded the module, then you need to look closely at your Asterisk log files because the module was not able to find any valid license files. ___ -- 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] opermode=?
On Thu, Apr 02, 2009 at 09:22:52AM -0700, bilal ghayyad wrote:
Hi All;
If I need to set the opermode to King Saudi Arabia, what the name I
have to use? For example, to set it for kuwait then I use
opermode=KUWAIT. So what will be for Saudi Arabia?
$ grep -i saudi drivers/dahdi/fxo_modes.h
{ .name = SAUDIARABIA,
--
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
___
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Re: [asterisk-users] Asterisk G729 codec...
asterisk*CLI module load codec_g729a.so asterisk*CLI [Apr 2 14:06:10] WARNING[3732]: loader.c:605 inspect_module: Module 'codec_g729a.so' does not provide a description. [Apr 2 14:06:10] WARNING[3732]: loader.c:662 load_resource: Module 'codec_g729a.so' could not be loaded. [Apr 2 14:06:10] WARNING[3732]: loader.c:605 inspect_module: Module 'codec_g729a.so' does not provide a description. [Apr 2 14:06:10] WARNING[3732]: loader.c:662 load_resource: Module 'codec_g729a.so' could not be loaded. What is this does not provide a description? I'm sure that I downloaded the codec from the digium page... On Thursday 02 April 2009 12:57:15 Miguel Molina wrote: criptos escribió: Humm... should the list would be magic again? I have just intsalled, using the register, benchmark and downloared the correct codec to my asterisk installation, but I don't have the g729 command at my CLI... Any advice... Do I reboot? ;D Did you load your brand new codec_g729.so module after putting it on /usr/lib/asterisk/modules? CLI module load codec_g729.so What does the command core show codecs show? Cheers, ___ -- 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] T1/PRI ignore answer signal
Hi Jerry, If you are calling that number from Asterisk via T1 and you do not get ANSWER/CONNECT message from that particular number/line then a workaround might not work. It's simply because there's no connectivity. You might only have early audio one way from that PA line to you but not the other way around. Martin On Thu, Apr 2, 2009 at 11:44 AM, Jerry Geis ge...@pagestation.com wrote: Is there anyway a T1/PRI can ignore the ANSWERED signal and just go straight from a dial command to the call was answered? I have a PBX that when calling a certain analog trunk it is not giving me signaling that the call was answered however I hear the PA system come off hook and give dial tone. So is there something in the dial command that can override the signaling and just go straight to the answered state so my AGI runs? The T1 is functioning normal for all other calls. Its just to this analog trunk that has the PA system connected to it. Jerry ___ -- 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] Asterisk G729 codec...
check if you loaded the module show modules like codec_g729 or simply try to unload/load codec_g729.so Martin On Thu, Apr 2, 2009 at 1:25 PM, criptos crip...@aullox.com wrote: Humm... should the list would be magic again? I have just intsalled, using the register, benchmark and downloared the correct codec to my asterisk installation, but I don't have the g729 command at my CLI... Any advice... Do I reboot? ;D ___ -- 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] Asterisk G729 codec...
criptos wrote: asterisk*CLI module load codec_g729a.so asterisk*CLI [Apr 2 14:06:10] WARNING[3732]: loader.c:605 inspect_module: Module 'codec_g729a.so' does not provide a description. [Apr 2 14:06:10] WARNING[3732]: loader.c:662 load_resource: Module 'codec_g729a.so' could not be loaded. [Apr 2 14:06:10] WARNING[3732]: loader.c:605 inspect_module: Module 'codec_g729a.so' does not provide a description. [Apr 2 14:06:10] WARNING[3732]: loader.c:662 load_resource: Module 'codec_g729a.so' could not be loaded. What is this does not provide a description? You are not using the proper version of codec_g729 for your version of Asterisk; there are different versions for different versions of Asterisk. Based on this message, I would suspect you are running some version of Asterisk 1.2, and thus you will need to download the codec_g729 module from the 'unsupported/asterisk-1.2' area of the downloads.digium.com site. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXO Ignore ring
I'd rather put Wait(3600) than Hangup(). Furthermore hangup would probably not work since the line was not taken offhook. Asterisk would do cleanup on the logical zap channel but then the next ring would create another zap channel and so on till the line stops ringing. Martin On Thu, Apr 2, 2009 at 10:49 AM, Marc Charbonneau timebandit...@gmail.com wrote: On Thu, Apr 2, 2009 at 11:37 AM, Cary Fitch ca...@usawide.net wrote: Is there a way to program an FXO device to totally ignore incoming calls? put the port in that context : [incoming-noanswer] exten = s,1,Hangup() hth ___ -- 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 Advice SIP 183 progessl
Do you know how to play a musiconhold or ... but when its ringing the user
will hear the Ring Back Tone
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Thursday, April 02, 2009 9:44 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Please Advice SIP 183 progessl
Try replacing answer() with playback(tt-monkeys)
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Khaled W.
Chehab
Sent: Thursday, April 02, 2009 1:33 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Please Advice SIP 183 progessl
I tried it but it didn't work even ,If I use Answer() function , Billing
will starts
Thanks
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Thursday, April 02, 2009 8:59 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Please Advice SIP 183 progessl
This is a hack-fix but if you Answer the call before dialing, that might
remove the progress message
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Khaled W.
Chehab
Sent: Thursday, April 02, 2009 12:47 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Please Advice SIP 183 progessl
Guys I tried the R,r options at the DIAL(SIP/,,rm) and also its sending
183
Any Advice
Dears
Kindly find my dial script below,I am trying to send the caller 180 ringing
but all tries were failed,
The caller always receive 183 session Progress
Even I add in the sip.conf
progressinband=never
exten = _X.,1,Wait(1)
exten = _X.,n,SetMusicOnHold(English)
exten = _X.,n,WaitMusicOnHold(2)
exten = _X.,n,NoOp(Return-)
-- add --
Exten = _X.,n,Answer()
-- end add --
exten = _X.,n,Dial(SIP/Gateway1/8430${EXTEN}|300|m)
;exten = _X.,n,SIPAddHeader(Alert-Info: Ring Answer)
exten = _X.,n,Goto(y-${DIALSTATUS},1) ; Jump based on status
(NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten = y-NOANSWER,1,SetMusicOnHold(busy)
exten = y-NOANSWER,n,WaitMusicOnHold(18) ; If busy, Playback Busy /
NOANSWER announce
exten = y-BUSY,1,SetMusicOnHold(busy)
exten = y-BUSY,n,WaitMusicOnHold(18) ; If busy, Playback Busy / busy
announce
exten = _y-.,1,Goto(y-NOANSWER,1) ; Treat anything else as no answer
exten = _X.,n,HangUp()
Please Advice
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Thursday, April 02, 2009 6:40 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] SIP 183 progessl
Sipaddheader(180 Ringing) might do the trick.
If you are compiling your own asterisk, you could change chan_sip.c to
replace 183 Session Progress with 180 Ringing (line 3950 in my source)
but that might break something else.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Khaled W.
Chehab
Sent: Thursday, April 02, 2009 10:23 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] SIP 183 progessl
Can you please tell me how to Custom SIP header
Regards
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Thursday, April 02, 2009 6:16 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Xorcom and Doorbell
Custom SIP header?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Khaled W.
Chehab
Sent: Thursday, April 02, 2009 10:02 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Xorcom and Doorbell
Dears
How can I send or force sending 180 Ringing instead of 183 back to the
caller ?or send both sequential if its impossible
I used progressinband=never but it did work .
Regards
*
No employee or agent is authorized to conclude any binding agreement on
behalf of Xplorium with another party by e-mail without express written
confirmation by an officer of Xplorium. Any views expressed by an individual
in this electronic message do not necessarily reflect views of Xplorium or
its subsidiaries and associates.
This electronic message and its attachments are solely addressed to the
addressee(s), and contain confidential information protected from disclosure
belonging to Xplorium.
If you are not the intended addressee of this electronic message and its
Re: [asterisk-users] Please Advice SIP 183 progessl
Kindly its too important to me
If any one can help me on a command can force asterisk to send 180 and 183
sip message in the same time
Regards
Do you know how to play a musiconhold or ... but when its ringing the user
will hear the Ring Back Tone
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Thursday, April 02, 2009 9:44 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Please Advice SIP 183 progessl
Try replacing answer() with playback(tt-monkeys)
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Khaled W.
Chehab
Sent: Thursday, April 02, 2009 1:33 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Please Advice SIP 183 progessl
I tried it but it didn't work even ,If I use Answer() function , Billing
will starts
Thanks
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Thursday, April 02, 2009 8:59 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Please Advice SIP 183 progessl
This is a hack-fix but if you Answer the call before dialing, that might
remove the progress message
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Khaled W.
Chehab
Sent: Thursday, April 02, 2009 12:47 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Please Advice SIP 183 progessl
Guys I tried the R,r options at the DIAL(SIP/,,rm) and also its sending
183
Any Advice
Dears
Kindly find my dial script below,I am trying to send the caller 180 ringing
but all tries were failed,
The caller always receive 183 session Progress
Even I add in the sip.conf
progressinband=never
exten = _X.,1,Wait(1)
exten = _X.,n,SetMusicOnHold(English)
exten = _X.,n,WaitMusicOnHold(2)
exten = _X.,n,NoOp(Return-)
-- add --
Exten = _X.,n,Answer()
-- end add --
exten = _X.,n,Dial(SIP/Gateway1/8430${EXTEN}|300|m)
;exten = _X.,n,SIPAddHeader(Alert-Info: Ring Answer)
exten = _X.,n,Goto(y-${DIALSTATUS},1) ; Jump based on status
(NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten = y-NOANSWER,1,SetMusicOnHold(busy)
exten = y-NOANSWER,n,WaitMusicOnHold(18) ; If busy, Playback Busy /
NOANSWER announce
exten = y-BUSY,1,SetMusicOnHold(busy)
exten = y-BUSY,n,WaitMusicOnHold(18) ; If busy, Playback Busy / busy
announce
exten = _y-.,1,Goto(y-NOANSWER,1) ; Treat anything else as no answer
exten = _X.,n,HangUp()
Please Advice
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Thursday, April 02, 2009 6:40 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] SIP 183 progessl
Sipaddheader(180 Ringing) might do the trick.
If you are compiling your own asterisk, you could change chan_sip.c to
replace 183 Session Progress with 180 Ringing (line 3950 in my source)
but that might break something else.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Khaled W.
Chehab
Sent: Thursday, April 02, 2009 10:23 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] SIP 183 progessl
Can you please tell me how to Custom SIP header
Regards
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Thursday, April 02, 2009 6:16 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Xorcom and Doorbell
Custom SIP header?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Khaled W.
Chehab
Sent: Thursday, April 02, 2009 10:02 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Xorcom and Doorbell
Dears
How can I send or force sending 180 Ringing instead of 183 back to the
caller ?or send both sequential if its impossible
I used progressinband=never but it did work .
Regards
*
No employee or agent is authorized to conclude any binding agreement on
behalf of Xplorium with another party by e-mail without express written
confirmation by an officer of Xplorium. Any views expressed by an individual
in this electronic message do not necessarily reflect views of Xplorium or
its subsidiaries and associates.
This electronic message and its attachments are solely addressed to the
addressee(s), and contain confidential
[asterisk-users] cant get a x100p works
I have an ubuntu 8.10 machine. Linux 2.6.27-14-generic i want to configure a x100p card an use it with asterisk, so i download, compile and install: asterisk-1.4.24 dahdi-linux-2.1.0.4 dahdi-tools-2.1.0.2 libpri-1.4.9 i try almost everything i found on the net but without success: if i run lspci: 04:06.0 Communication controller: Motorola Wildcard X100P when i run dahdi_hardware appears this: pci::04:06.0 wcfxo- 1057:5608 Wildcard X100P if i run dahdi_cfg -v : DAHDI Tools Version - 2.1.0.2 DAHDI Version: 2.1.0.4 Echo Canceller(s): Configuration == 0 channels to configure. when i run dahdi_scan: [1] active=yes alarms=UNCONFIGURED description=DAHDI_DUMMY/1 (source: HRtimer) 1 name=DAHDI_DUMMY/1 manufacturer= devicetype=DAHDI Dummy Timing location= basechan=1 totchans=0 irq=0 if i fo dahdigenconf everythink still same. I also reboot and do modprobe wcfxo. not success... ___ -- 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] problema con una x100p
Tengo en una maquia ubuntu 8.10 el kernel es Linux 2.6.27-14-generic Quiero configurar una tarjeta x100p i usarla con asterisk, asi que descague compile e instale lo siguiente: asterisk-1.4.24 dahdi-linux-2.1.0.4 dahdi-tools-2.1.0.2 libpri-1.4.9 Sin embargo no logro configurar la tarjeta con exito, ya probe casi todo. Esto aparece si ejecuto lspci: 04:06.0 Communication controller: Motorola Wildcard X100P dahdi_hardware me muestra: pci::04:06.0 wcfxo- 1057:5608 Wildcard X100P dahdi_cfg -v : DAHDI Tools Version - 2.1.0.2 DAHDI Version: 2.1.0.4 Echo Canceller(s): Configuration == 0 channels to configure. dahdi_scan: [1] active=yes alarms=UNCONFIGURED description=DAHDI_DUMMY/1 (source: HRtimer) 1 name=DAHDI_DUMMY/1 manufacturer= devicetype=DAHDI Dummy Timing location= basechan=1 totchans=0 irq=0 Nada parece funcionar y realmente no se donde esta el error... alguna idea? ___ -- 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] 2-3 Calls at a time
Many time we face an issue where even if an agent is on Call, another call
comes in.
Sometimes, even if agent hang up the call, call stays back and another come
sin and then both customers can hear each other { which i think is VERY
dangerous [image: Wink] }
Any Solutions ?
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[asterisk-users] FXS Line Voltage When Dahdi/Zaptel is off?
Hi - Does anybody know if an FXS generates line voltage when Dahdi/Zaptel is disabled? Thanks, Noah ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] cant get a x100p works
On Thu, Apr 02, 2009 at 03:51:21PM -0500, Manolet Gmail wrote: I have an ubuntu 8.10 machine. Linux 2.6.27-14-generic i want to configure a x100p card an use it with asterisk, so i download, compile and install: asterisk-1.4.24 dahdi-linux-2.1.0.4 dahdi-tools-2.1.0.2 libpri-1.4.9 i try almost everything i found on the net but without success: if i run lspci: 04:06.0 Communication controller: Motorola Wildcard X100P when i run dahdi_hardware appears this: pci::04:06.0 wcfxo- 1057:5608 Wildcard X100P What's the output of: lsmod | grep ^dahdi -- 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] Anyone actually built h323plus on Fedora?
I've been trying to build h323plus (both the release and svn) for chan_h323 on Fedora 10. No joy. I posted on the h323plus ml, but no response. Anybody here actually built it on Fedora? Wanna share your secrets, or even better a specfile? sean ___ -- 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] problema con una x100p
nos muestran la configuración de sus líneas de /etc/dahdi/system.conf y /etc/asterisk/chan/dahdi.conf 2009/4/2 Manolet Gmail mano...@gmail.com Tengo en una maquia ubuntu 8.10 el kernel es Linux 2.6.27-14-generic Quiero configurar una tarjeta x100p i usarla con asterisk, asi que descague compile e instale lo siguiente: asterisk-1.4.24 dahdi-linux-2.1.0.4 dahdi-tools-2.1.0.2 libpri-1.4.9 Sin embargo no logro configurar la tarjeta con exito, ya probe casi todo. Esto aparece si ejecuto lspci: 04:06.0 Communication controller: Motorola Wildcard X100P dahdi_hardware me muestra: pci::04:06.0 wcfxo- 1057:5608 Wildcard X100P dahdi_cfg -v : DAHDI Tools Version - 2.1.0.2 DAHDI Version: 2.1.0.4 Echo Canceller(s): Configuration == 0 channels to configure. dahdi_scan: [1] active=yes alarms=UNCONFIGURED description=DAHDI_DUMMY/1 (source: HRtimer) 1 name=DAHDI_DUMMY/1 manufacturer= devicetype=DAHDI Dummy Timing location= basechan=1 totchans=0 irq=0 Nada parece funcionar y realmente no se donde esta el error... alguna idea? ___ -- 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] Simple Queue question
Hello, I have a fairly standard call center. I'm running 1.4.23.1. I am trying to get a mechanism where : 1- Employee A can have the phone at his desk ring when a call comes in the queue 2- When already on a call, employee A does not hear a beep in his phone because another call is trying to come in It's fairly simple. I tried a few different things: in queues.conf [559] member = Agent/109988 The issue is the that the agent needs to wait on the phone for a call to come in. I read http://www.voip-info.org/wiki/view/Asterisk+cmd+AgentCallbackLogin but it will be deprecated and the doc/queues-with-callback-members.txt means that I would have to convert to AEL (unless I can do extensions.conf and extensions.ael at the same time. Not sure) I tried this as well: [559] member = Local/6...@q2a/n member = Local/7...@q2a/n In conjunction with http://www.voip-info.org/wiki/view/Asterisk+Queue+with+limited+calls+per+IAX+agent . It's been causing issues where an incoming call will go in the queue and even though I have staff ready to answer, the system does not ring physically ring anyone (the logs show different, therefore huge confusion) I am guessing many of you here on this list are doing or have done something fairly close to what I am attempting. I welcome your help. Haim. ___ -- 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] Simple Queue question
On Thu, 2 Apr 2009, Haim Dimer wrote: The issue is the that the agent needs to wait on the phone for a call to come in. I read http://www.voip-info.org/wiki/view/Asterisk+cmd+AgentCallbackLogin but it will be deprecated and the doc/queues-with-callback-members.txt means that I would have to convert to AEL (unless I can do extensions.conf and extensions.ael at the same time. Not sure) I'm a 1.2 Luddite, but you can use both extensions.conf and extensions.ael. You can load an ael and do a show dialplan to see how Asterisk converts AEL to conf. Thanks in advance, Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- 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] cant get a x100p works
On Thu, Apr 2, 2009 at 4:38 PM, Tzafrir Cohen tzafrir.co...@xorcom.com wrote: On Thu, Apr 02, 2009 at 03:51:21PM -0500, Manolet Gmail wrote: I have an ubuntu 8.10 machine. Linux 2.6.27-14-generic i want to configure a x100p card an use it with asterisk, so i download, compile and install: asterisk-1.4.24 dahdi-linux-2.1.0.4 dahdi-tools-2.1.0.2 libpri-1.4.9 i try almost everything i found on the net but without success: if i run lspci: 04:06.0 Communication controller: Motorola Wildcard X100P when i run dahdi_hardware appears this: pci::04:06.0 wcfxo- 1057:5608 Wildcard X100P What's the output of: lsmod | grep ^dahdi r...@lhserver:~# lsmod | grep ^dahdi dahdi_dummy11620 0 dahdi_transcode15244 1 wctc4xxp dahdi 202280 13 dahdi_dummy,xpp,dahdi_transcode,wcb4xxp,wctdm,wcfxo,wctdm24xxp,wcte11xp,wct1xxp,wcte12xp,wct4xxp ___ -- 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] problema con una x100p
system.conf: # Global data loadzone= us defaultzone = us el archivo /etc/asterisk/chan/dahdi.conf no existe, estoy usando asterisk 1.2 de todas maneras como dije ztcfg -v tampoco muestra la tarjeta, asi que de ninguna forma la va a ver asterisk, si hago en asterisk dahdi show channels no aparece nada. 2009/4/2 Brandon B. bran...@brellsystems.com: nos muestran la configuración de sus líneas de /etc/dahdi/system.conf y /etc/asterisk/chan/dahdi.conf 2009/4/2 Manolet Gmail mano...@gmail.com Tengo en una maquia ubuntu 8.10 el kernel es Linux 2.6.27-14-generic Quiero configurar una tarjeta x100p i usarla con asterisk, asi que descague compile e instale lo siguiente: asterisk-1.4.24 dahdi-linux-2.1.0.4 dahdi-tools-2.1.0.2 libpri-1.4.9 Sin embargo no logro configurar la tarjeta con exito, ya probe casi todo. Esto aparece si ejecuto lspci: 04:06.0 Communication controller: Motorola Wildcard X100P dahdi_hardware me muestra: pci::04:06.0 wcfxo- 1057:5608 Wildcard X100P dahdi_cfg -v : DAHDI Tools Version - 2.1.0.2 DAHDI Version: 2.1.0.4 Echo Canceller(s): Configuration == 0 channels to configure. dahdi_scan: [1] active=yes alarms=UNCONFIGURED description=DAHDI_DUMMY/1 (source: HRtimer) 1 name=DAHDI_DUMMY/1 manufacturer= devicetype=DAHDI Dummy Timing location= basechan=1 totchans=0 irq=0 Nada parece funcionar y realmente no se donde esta el error... alguna idea? ___ -- 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] async agi question
Yes, I have the patch around here. I think it's the one you said at http://www.moythreads.com/wordpress/2007/12/24/asterisk-asynchronous-agi/ Due to the res_agi patch excedes the size limit for this mailing list, (40Kb) I wasn't able to attach it on this post, so you can find it at http://docs.google.com/Doc?id=ddb4rkts_0fd9z5qcr Thanks Jose 2009/4/2 Moises Silva Async AGI was never released for Asterisk 1.4.X, so probably the patch you used has a bug or something, do you still have the patch around? Moy ___ -- 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] Ring group howto
How do I manually set up a ring group? All the info I've Googled tells me how to do this using Trixbox or FreePBX. I am using standard Asterisk 1.4 configuring at the CLI. Michael ___ -- 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] problema con una x100p
Follow instructions from the following line to configure Asterisk 1.2 with zaptel drivers for the X100P. If you are using dahdi-linux drivers instead of zaptel, then instead of zaptel.conf you need to have a properly configured /etc/dahdi/system.conf and instead of /etc/asterisk/zapata.conf use this file instead /etc/asterisk/chan_dahdi.conf. http://users.telenet.be/Asterisk-PBX/InstallWildcard.htm Since it sounds like you have not used Asterisk before, you should read Asterisk: the future of telephony http://astbook.asteriskdocs.org/ and look through the pages at voip-info.org to get started. 2009/4/2 Manolet Gmail mano...@gmail.com system.conf: # Global data loadzone= us defaultzone = us el archivo /etc/asterisk/chan/dahdi.conf no existe, estoy usando asterisk 1.2 de todas maneras como dije ztcfg -v tampoco muestra la tarjeta, asi que de ninguna forma la va a ver asterisk, si hago en asterisk dahdi show channels no aparece nada. 2009/4/2 Brandon B. bran...@brellsystems.com: nos muestran la configuración de sus líneas de /etc/dahdi/system.conf y /etc/asterisk/chan/dahdi.conf 2009/4/2 Manolet Gmail mano...@gmail.com Tengo en una maquia ubuntu 8.10 el kernel es Linux 2.6.27-14-generic Quiero configurar una tarjeta x100p i usarla con asterisk, asi que descague compile e instale lo siguiente: asterisk-1.4.24 dahdi-linux-2.1.0.4 dahdi-tools-2.1.0.2 libpri-1.4.9 Sin embargo no logro configurar la tarjeta con exito, ya probe casi todo. Esto aparece si ejecuto lspci: 04:06.0 Communication controller: Motorola Wildcard X100P dahdi_hardware me muestra: pci::04:06.0 wcfxo- 1057:5608 Wildcard X100P dahdi_cfg -v : DAHDI Tools Version - 2.1.0.2 DAHDI Version: 2.1.0.4 Echo Canceller(s): Configuration == 0 channels to configure. dahdi_scan: [1] active=yes alarms=UNCONFIGURED description=DAHDI_DUMMY/1 (source: HRtimer) 1 name=DAHDI_DUMMY/1 manufacturer= devicetype=DAHDI Dummy Timing location= basechan=1 totchans=0 irq=0 Nada parece funcionar y realmente no se donde esta el error... alguna idea? ___ -- 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 ___ -- 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] Ring group howto
A group of phones that ring all at once? Like: exten = 5226001454,1,Dial(SIP/3615221401SIP/3615221402SIP/3615221407SIP/522600140 5,20) Take out the line breaks. Or were you looking for something else? CF -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michael Sent: Thursday, April 02, 2009 6:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Ring group howto How do I manually set up a ring group? All the info I've Googled tells me how to do this using Trixbox or FreePBX. I am using standard Asterisk 1.4 configuring at the CLI. Michael ___ -- 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] Ring group howto
On Fri, 03 Apr 2009 12:32:03 you wrote: A group of phones that ring all at once? Like: exten = 5226001454,1,Dial(SIP/3615221401SIP/3615221402SIP/3615221407SIP/52260014 0 5,20) Take out the line breaks. Or were you looking for something else? CF That is what I am currently doing - though is there a cleaner way? ___ -- 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, TE220 Device, and Asterisk Problem
Hello! I am trying to configure my digium TE220 dual-span pci express card with Dahdi. I seemed to have managed to set up the card with the Dahdi kernel, as demonstrated by executing dahdi_scan: [1] active=yes alarms=RED description=T2XXP (PCI) Card 0 Span 1 name=TE2/0/1 manufacturer=Digium devicetype=Wildcard TE220 (4th Gen) location=Board ID Switch 0 basechan=1 totchans=31 irq=16 type=digital-E1 syncsrc=0 lbo=0 db (CSU)/0-133 feet (DSX-1) coding_opts=HDB3 framing_opts=CCS,CRC4 coding=HDB3 framing=CCS [2] active=yes alarms=RED description=T2XXP (PCI) Card 0 Span 2 name=TE2/0/2 manufacturer=Digium devicetype=Wildcard TE220 (4th Gen) location=Board ID Switch 0 basechan=32 totchans=31 irq=16 type=digital-E1 syncsrc=0 lbo=0 db (CSU)/0-133 feet (DSX-1) coding_opts=HDB3 framing_opts=CCS,CRC4 coding=HDB3 framing=CCS However, when calling a pri show spans in asterisk, nothing comes up. Although the physical cables are not yet connected, shouldn't it state something like: PRI span 1/0: Provisioned, In Alarm, Down, Active PRI span 2/0: Provisioned, In Alarm, Down, Active Any help will be appreciated! 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] 2-3 Calls at a time
Is that a Bug in asterisk and meetme file ?
On Fri, Apr 3, 2009 at 2:27 AM, David @ULC ucoms2...@gmail.com wrote:
Many time we face an issue where even if an agent is on Call, another call
comes in.
Sometimes, even if agent hang up the call, call stays back and another come
sin and then both customers can hear each other { which i think is VERY
dangerous [image: Wink] }
Any Solutions ?
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Re: [asterisk-users] cant get a x100p works
On Thu, Apr 02, 2009 at 05:55:04PM -0500, Manolet Gmail wrote: On Thu, Apr 2, 2009 at 4:38 PM, Tzafrir Cohen tzafrir.co...@xorcom.com wrote: On Thu, Apr 02, 2009 at 03:51:21PM -0500, Manolet Gmail wrote: I have an ubuntu 8.10 machine. Linux 2.6.27-14-generic i want to configure a x100p card an use it with asterisk, so i download, compile and install: asterisk-1.4.24 dahdi-linux-2.1.0.4 dahdi-tools-2.1.0.2 libpri-1.4.9 i try almost everything i found on the net but without success: if i run lspci: 04:06.0 Communication controller: Motorola Wildcard X100P when i run dahdi_hardware appears this: pci::04:06.0 wcfxo- 1057:5608 Wildcard X100P What's the output of: lsmod | grep ^dahdi r...@lhserver:~# lsmod | grep ^dahdi dahdi_dummy11620 0 dahdi_transcode15244 1 wctc4xxp dahdi 202280 13 dahdi_dummy,xpp,dahdi_transcode,wcb4xxp,wctdm,wcfxo,wctdm24xxp,wcte11xp,wct1xxp,wcte12xp,wct4xxp Could you please run the following command and provide their output? rmmod wcfxo modprobe wcfxo dmesg | tail -- 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
Re: [asterisk-users] Dahdi, TE220 Device, and Asterisk Problem
On Thu, Apr 2, 2009 at 4:36 PM, Elliot Murdock murdo...@gmail.com wrote: Hello! I am trying to configure my digium TE220 dual-span pci express card with Dahdi. I seemed to have managed to set up the card with the Dahdi kernel, as demonstrated by executing dahdi_scan: [1] active=yes alarms=RED description=T2XXP (PCI) Card 0 Span 1 name=TE2/0/1 manufacturer=Digium devicetype=Wildcard TE220 (4th Gen) location=Board ID Switch 0 basechan=1 totchans=31 irq=16 type=digital-E1 syncsrc=0 lbo=0 db (CSU)/0-133 feet (DSX-1) coding_opts=HDB3 framing_opts=CCS,CRC4 coding=HDB3 framing=CCS [2] active=yes alarms=RED description=T2XXP (PCI) Card 0 Span 2 name=TE2/0/2 manufacturer=Digium devicetype=Wildcard TE220 (4th Gen) location=Board ID Switch 0 basechan=32 totchans=31 irq=16 type=digital-E1 syncsrc=0 lbo=0 db (CSU)/0-133 feet (DSX-1) coding_opts=HDB3 framing_opts=CCS,CRC4 coding=HDB3 framing=CCS However, when calling a pri show spans in asterisk, nothing comes up. Although the physical cables are not yet connected, shouldn't it state something like: PRI span 1/0: Provisioned, In Alarm, Down, Active PRI span 2/0: Provisioned, In Alarm, Down, Active Any help will be appreciated! 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 Silly question but did you install libpri? If so what version? ___ -- 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] cant get a x100p works
Then you need to edit /etc/dahdi/system.conf manually and add fxsks=1 then dahdi_cfg -vv then check if wcfxo module takes interrupts dahdi_test Martin On Thu, Apr 2, 2009 at 5:55 PM, Manolet Gmail mano...@gmail.com wrote: What's the output of: lsmod | grep ^dahdi r...@lhserver:~# lsmod | grep ^dahdi dahdi_dummy 11620 0 dahdi_transcode 15244 1 wctc4xxp dahdi 202280 13 dahdi_dummy,xpp,dahdi_transcode,wcb4xxp,wctdm,wcfxo,wctdm24xxp,wcte11xp,wct1xxp,wcte12xp,wct4xxp ___ -- 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, TE220 Device, and Asterisk Problem
maybe you have to call the dahdi_scan with some argument to autogenerate config files ... for any dahdi T1/E1 card you have to work properly you have to have /etc/dahdi/system.conf configured with span= and bchan= and dchan= keyword and /etc/asterisk/dahdi*.conf with channel = keyword. check it out ... this is the first step... dahdi_cfg -vv should show all your 64 channels Martin On Thu, Apr 2, 2009 at 6:36 PM, Elliot Murdock murdo...@gmail.com wrote: Hello! I am trying to configure my digium TE220 dual-span pci express card with Dahdi. I seemed to have managed to set up the card with the Dahdi kernel, as demonstrated by executing dahdi_scan: [1] active=yes alarms=RED description=T2XXP (PCI) Card 0 Span 1 name=TE2/0/1 manufacturer=Digium devicetype=Wildcard TE220 (4th Gen) location=Board ID Switch 0 basechan=1 totchans=31 irq=16 type=digital-E1 syncsrc=0 lbo=0 db (CSU)/0-133 feet (DSX-1) coding_opts=HDB3 framing_opts=CCS,CRC4 coding=HDB3 framing=CCS [2] active=yes alarms=RED description=T2XXP (PCI) Card 0 Span 2 name=TE2/0/2 manufacturer=Digium devicetype=Wildcard TE220 (4th Gen) location=Board ID Switch 0 basechan=32 totchans=31 irq=16 type=digital-E1 syncsrc=0 lbo=0 db (CSU)/0-133 feet (DSX-1) coding_opts=HDB3 framing_opts=CCS,CRC4 coding=HDB3 framing=CCS However, when calling a pri show spans in asterisk, nothing comes up. Although the physical cables are not yet connected, shouldn't it state something like: PRI span 1/0: Provisioned, In Alarm, Down, Active PRI span 2/0: Provisioned, In Alarm, Down, Active Any help will be appreciated! 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
