[asterisk-users] Problem in Detecting Dtmf on FXO line.
Hi All, I am having Problem in detecting DTMF on analog lines. basically are system is in india and telco provider is BSNL [Bharat sanchar Nigam LImited]. We have Purchased Analog card From chinaroby.com which is X1600P 16 port FXO card. they also provide us wctdm.c file. card is detected successfully, incoming and outgoing calls scenario is also fine. we are unable to receive dtmf properly it means there is some digit are missing when we receive dtmf the ratio of sucess is about to 70% and 30% of calls are getting wrong dtmf . Dahdi version is 2.3.0.1 and asterisk version 1.6.0.24 I load module using modprobe wctdm opermod=INDIA cidbeforering=1 cidbuflen=1 fixedtimepolarity=16 here id chan_dahdi.conf. [trunkgroups] [channels] context=from-zaptel signalling=fxs_ks busydetect=yes busycount=4 ;rxwink=300 ; Atlas seems to use long (250ms) winks usecallerid=yes callerid=asreceived cidstart=polarity_in cidsignalling=dtmf hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes callprogess=yes echocancel=yes echocancelwhenbridged=no echotraining=800 rxgain=0.0 txgain=0.0 ;cid_rxgain=5.0 relaxdtmf=yes callgroup=1 pickupgroup=1 toneduration=500 ;answeronpolarityswitch=yes hanguponpolarityswitch=yes ;polarityonanswerdelay=1000 group=0 channel => 1 ;channel => 2 ;channel => 3 ;channel => 4 ;channel => 5 ;channel => 6 ;channel => 7 ;channel => 8 ;channel => 9 ;channel => 10 ;channel => 11 ;channel => 12 ;channel => 13 ;channel => 14 ;channel => 15 ;channel => 16 Also set tonezone = in in system.conf, tried many solutions and changed so many parameters of chan_dahdi.cong but still i am not getting successful result. Please share your comments if anyone have idea for india specific region . Regards Dhaval -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom Blf / Directed Pickup
2011/7/7 Gord Urquhart > Oliver >Your problem is you have not turned on "notifycid=yes" in sip.conf. Back > on June 28 in another thread you said > > "With asterisk 1.6.1.18, I could make this work without setting > notifycid=yes isn sip.conf." > > butyes that gets the monitored line to blink on an incoming call, but > as you have discovered the phone will not do a directed pickup. This info is > also available at > http://www.voip-info.org/wiki/view/Asterisk+presence > > cheers > gord > In my previous checks, I concluded notifycid option was not supported in Asterisk 1.6.1.X. So I wondered if others could like I did, have confused a General Pickup with which Asterisk receives a the string enabled in features.conf, with a Directed Pickup with which Asterisk receives a full INVITE which is treated according the dialplan. I sure need to double-check that. Thanks for your reply. > > On Wed, Jul 6, 2011 at 8:22 AM, Olivier wrote: > >> Using a Polycom 650 with 3.3.1, I could not have Directed Pickup working. >> >> More precisely, I configured the phone using and >> entries as described in this thread. >> Whenever a call comes in, BLF is blinking green. >> Pressing the associated key generate generates a general Call Pickup (*8), >> not a directed Call Pickup. >> >> Could you confirm this ? >> >> Regards >> >> -- >> _ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Eyebeam crashes when dialing an invalid number...
As asterisk is an B2BUA you can handle 503 at asterisk and hang caller end using the response code compatible with eyebeam as Hangup(16) Regards, Faisal -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Carlos Chavez Sent: Thursday, July 07, 2011 10:17 PM To: Asterisk Subject: [asterisk-users] Eyebeam crashes when dialing an invalid number... Lately I have been getting many complains that Eyebeam crashes when you dial a number that does not exist. This happens in both R2 and ISDN PRI lines. The softphone stops working and has to be restarted. The response I got from tech support was: the actual issue is that asterisk should not be sending a 503 service unavailable when a particular softphone is not online. The soft phone stops because a 503 means that the server itself is unavailable. Does anyone have a workaround for this? Maybe a way to manipulate via dialplan so the softphone does not get the 503 message? -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Stripping characters from ${CALLERID(num)} ?
Use Filter command in dia-plan to get numeric only string,
Set(MYNEWCLI=${FILTER(0123456789,${CALLERID(number)})
Regards,
Faisal
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tim Nelson
Sent: Thursday, July 07, 2011 9:44 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Stripping characters from ${CALLERID(num)} ?
Greetings-
On occasion, I have calls coming into an Asterisk 1.2.x system where the
${CALLERID(num)} includes '-'. Ex:
123-456-7890
How can I strip the dashes from the number, leaving me with '1234567890'?
I've tried the following which does not appear to be working:
Dialplan:
exten => _X.,n,Set(PROPERCID=System(echo ${CALLERID(num)} | sed s/\-//g))
exten => _X.,n,NoOp(Fixed proper CID is ${PROPERCID}
Console Output:
-- Executing [11@cidmangletest:4] Set("SIP/w.x.y.z-b4d55ce8",
"PROPERCID=System(echo 123-456-7890 | sed s/\-//g)")
-- Executing [11@cidmangletest:5] NoOp("SIP/w.x.y.z-b4d55ce8",
"Fixed proper CID is System(echo 123-456-7890 | sed s/-//g)")
Obviously, I'm trying to throw the CID through sed via System() to strip the
dashes. Can anyone explain how to accomplish this? Or even better yet, how
to strip the dashes directly in the dialplan without the use of System()?
Thanks!
--Tim
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Re: [asterisk-users] Anybody doing PRI over IP?
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tim Nelson Sent: Thursday, July 07, 2011 4:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Anybody doing PRI over IP? - Original Message - > There is a "T1 over Ethernet" scheme that runs a T1 over Ethernet, Up > all the time. It consumes 1.5 +/- megs 24/7. > > I would suspect that was what was being offered. > Wow, that sounds horrifically inefficient. *MAYBE* keep the D channel open all the time if necessary, but what about all the channels that are not in use? Just waste 64k for each channel just because? Do you have any information on this technology, or the name of the vendor that offers it? You've piqued my interest. :) --Tim The reason for this scheme as proposed to us was for FAX use rather than other means. The people who proposed it however within a week converted to "Plan B". So, I have no further info. CF -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anybody doing PRI over IP?
- Original Message - > > -Original Message- > > From: asterisk-users-boun...@lists.digium.com > > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of > > Tim Nelson > > Sent: Thursday, July 07, 2011 4:18 PM > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Subject: Re: [asterisk-users] Anybody doing PRI over IP? > > > > - Original Message - > > > There is a "T1 over Ethernet" scheme that runs a T1 over > > Ethernet, Up > > > all the time. It consumes 1.5 +/- megs 24/7. > > > > > > I would suspect that was what was being offered. > > > > > > > Wow, that sounds horrifically inefficient. *MAYBE* keep the D > > channel open all the time if necessary, but what about all > > the channels that are not in use? Just waste 64k for each > > channel just because? > > I work for a CLEC. For our VoIP service we offer the customer a choice > of how we handoff to the customer. Customers can pick POTS, PRI, CT1, > or SIP handoff. Calls are SIP from our IAD/CPE device at the customer > (usually an Adtran box w/SIP) all the way to our core. Many carriers > do this, there is nothing special about it. > Right, this is how I expected it to operate. My prior question though was regarding the 'T1 over Ethernet' scheme someone mentioned which ran full throughput all the time. --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anybody doing PRI over IP?
> -Original Message- > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of > Tim Nelson > Sent: Thursday, July 07, 2011 4:18 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Anybody doing PRI over IP? > > - Original Message - > > There is a "T1 over Ethernet" scheme that runs a T1 over > Ethernet, Up > > all the time. It consumes 1.5 +/- megs 24/7. > > > > I would suspect that was what was being offered. > > > > Wow, that sounds horrifically inefficient. *MAYBE* keep the D > channel open all the time if necessary, but what about all > the channels that are not in use? Just waste 64k for each > channel just because? I work for a CLEC. For our VoIP service we offer the customer a choice of how we handoff to the customer. Customers can pick POTS, PRI, CT1, or SIP handoff. Calls are SIP from our IAD/CPE device at the customer (usually an Adtran box w/SIP) all the way to our core. Many carriers do this, there is nothing special about it. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anybody doing PRI over IP?
You can try digium. They provide the T1 cards. Check http://www.digium.com/ Thanks Indranil -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tim Nelson Sent: Thursday, July 07, 2011 4:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Anybody doing PRI over IP? - Original Message - > There is a “T1 over Ethernet” scheme that runs a T1 over Ethernet, Up > all the time. It consumes 1.5 +/- megs 24/7. > > I would suspect that was what was being offered. > Wow, that sounds horrifically inefficient. *MAYBE* keep the D channel open all the time if necessary, but what about all the channels that are not in use? Just waste 64k for each channel just because? Do you have any information on this technology, or the name of the vendor that offers it? You've piqued my interest. :) --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anybody doing PRI over IP?
- Original Message - > There is a “T1 over Ethernet” scheme that runs a T1 over Ethernet, Up > all the time. It consumes 1.5 +/- megs 24/7. > > I would suspect that was what was being offered. > Wow, that sounds horrifically inefficient. *MAYBE* keep the D channel open all the time if necessary, but what about all the channels that are not in use? Just waste 64k for each channel just because? Do you have any information on this technology, or the name of the vendor that offers it? You've piqued my interest. :) --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anybody doing PRI over IP?
There is a "T1 over Ethernet" scheme that runs a T1 over Ethernet, Up all the time. It consumes 1.5 +/- megs 24/7. I would suspect that was what was being offered. Cary _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bryant Zimmerman Sent: Thursday, July 07, 2011 4:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Anybody doing PRI over IP? _ From: "eric weaver" Sent: Thursday, July 07, 2011 4:41 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Anybody doing PRI over IP? A carrier I like will be introducing PRI over IP, presumably going thru some sort of gateway box (I'm guessing by Adtran but no data yet). Has anybody set up successfully to work directly with such a feed without bothering to take it down to T1 and use a T1/PRI card? Thanks eric I agree with others that likely what you are getting is a product that is SIP based and it is just being priced and bundled to compete with a PRI connection as most bussiness owners and phone guys know what a PRI is.. We have pri's into gateways that run on our VOIP network and we have sip trunks and we mix services out to our customers based on what the routes require. Most of our up line CLEC's can now deliver their TDM and SIP services in both forms so in most cases we take the SIP version and where the vendor does not support SIP correctly we take their PRI version and convert it to SIP ourselves on our gateways. zktech -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Stripping characters from ${CALLERID(num)} ?
On Thu, 7 Jul 2011, Tim Nelson wrote: In fact, I've noticed many of your posts are Asterisk 1.2.x and AGI/C oriented which is very unique. I confess. I'm a 1.2 Luddite. I know a little C, but would likely start out using PHPAGI as it's more familiar to me. I know, not as efficient, but a stepping stone. Fortunately, C is my sharpest tool. The problem with scripting languages is that you can execute hundreds of AGIs written in C in the time it takes the interpreter to load and parse your script. Another problem is that scripting languages don't complain about syntax errors until they are executed. If you mung an infrequently executed section of code, you may not find the error for a long time. The C compiler finds most of my fat-fingered accidents for me. You can go the 'fastagi' route and write 'scripted daemons' to handle your agi tasks, but then you are introducing a new level of complexity and new failure points. I'm currently running some database code in dialplan and its *NASTY*. If anything, DB access would be the incentive for me to dive into AGI. I confess I've never done any DB code in dialplan beyond a simple select. I've looked at what others have done and the hoops they have to crawl through to achieve an inferior solution and decided I should stick to what I know. -- 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anybody doing PRI over IP?
From: "eric weaver" Sent: Thursday, July 07, 2011 4:41 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Anybody doing PRI over IP? A carrier I like will be introducing PRI over IP, presumably going thru some sort of gateway box (I'm guessing by Adtran but no data yet). Has anybody set up successfully to work directly with such a feed without bothering to take it down to T1 and use a T1/PRI card? Thanks eric I agree with others that likely what you are getting is a product that is SIP based and it is just being priced and bundled to compete with a PRI connection as most bussiness owners and phone guys know what a PRI is.. We have pri's into gateways that run on our VOIP network and we have sip trunks and we mix services out to our customers based on what the routes require. Most of our up line CLEC's can now deliver their TDM and SIP services in both forms so in most cases we take the SIP version and where the vendor does not support SIP correctly we take their PRI version and convert it to SIP ourselves on our gateways. zktech -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anybody doing PRI over IP?
- Original Message - > A carrier I like will be introducing PRI over IP, presumably going > thru some sort of gateway box (I'm guessing by Adtran but no data > yet). Has anybody set up successfully to work directly with such a > feed without bothering to take it down to T1 and use a T1/PRI card? > Since you're getting this delivered via IP, I assume you already have 'Internet connectivity' of some sort. So, this "PRI over IP" is likely for voice? In most instances, this just means the carrier is giving you 23 simultaneous channels (or fractional) of VoIP connectivity and calling it a "PRI" for marketing speak. Some examples: http://www.ipcomms.net/html/package-virtualt1.html http://www.didlive.com/virtual_t1.htm I'm not saying this is a bad thing, just that it really isn't anything 'groundbreaking' or 'special'. I *have* used such a service before. In one case, the 'virtual PRI' was terminated to me via SIP via my Asterisk PBX box. In another instance, it was handed off via SIP to a Cisco gateway which presented a standard PRI port to the customer PBX equipment. In either case, the general idea is that you already have IP transport, this is for voice, and the channels are provided by SIP. Your termination equipment could likely be anything that handles SIP. If this turns out to be something *not* disguised as just "SIP in X number of channels form", I'd be interested to hear details. Maybe some sort of TDMoIP service? --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anybody doing PRI over IP?
On 07/07/2011 04:41 PM, eric weaver wrote: A carrier I like will be introducing PRI over IP, presumably going thru some sort of gateway box (I'm guessing by Adtran but no data yet). Has anybody set up successfully to work directly with such a feed without bothering to take it down to T1 and use a T1/PRI card? Are you talking about a TDMoIP solution? Or are you talking about trunking calls over an IP medium with PRI as the last-mile handoff at both ends? -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Anybody doing PRI over IP?
A carrier I like will be introducing PRI over IP, presumably going thru some sort of gateway box (I'm guessing by Adtran but no data yet). Has anybody set up successfully to work directly with such a feed without bothering to take it down to T1 and use a T1/PRI card? Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Stripping characters from ${CALLERID(num)} ?
From: "Bryant Zimmerman"
Sent: Thursday, July 07, 2011 4:14 PM
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Subject: Re: [asterisk-users] Stripping characters from ${CALLERID(num)} ?
Here is a simple way to strip the '-'
Here is a concept solution.
I have not tested the code so there may be some syntax errors.
It can work as I am doing stuff like this all the time. This example is
using a check to only do the cut if there is more than one field. You may
be able to just use step 3 from ctx and have what you want, but I am not
sure if it will fall back gracefully if there is only 1 field.
No AGI required.
exten => mycode,n,Gosub(ctx,1)
exten => ctx,1,Set(l_filedCNT=${FIELDQTY(CALLERID(num),-)})
exten => ctx,n,GotoIf($[${MATH(${l_filedCNT}>1)}=TRUE]?DoStrip:doSkip)
exten => ctx,n(doStrip),Set (CALLERID(num)=${CUT(CALLERID(num), -, 1-)})
exten => ctx,n(doSkip),NoOp(${CALLERID(num)})
Thanks
zktech
Made a correct in the above to reflect the skip state doSkip vs SkpStrip
and made syntax correct to the Set/CUT line. There may still be a few more
syntax issues in there.
zktech
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Re: [asterisk-users] Stripping characters from ${CALLERID(num)} ?
Here is a simple way to strip the '-'
Here is a concept solution.
I have not tested the code so there may be some syntax errors.
It can work as I am doing stuff like this all the time. This example is
using a check to only do the cut if there is more than one field. You may
be able to just use step 3 from ctx and have what you want, but I am not
sure if it will fall back gracefully if there is only 1 field.
No AGI required.
exten => mycode,n,Gosub(ctx,1)
exten => ctx,1,Set(l_filedCNT=${FIELDQTY(CALLERID(num),-)})
exten => ctx,n,GotoIf($[${MATH(${l_filedCNT}>1)}=TRUE]?DoStrip:SkipStrip)
exten => ctx,n(doStrip),Set CALLERID(num)=${CUST(CALLERID(num), -, 1-)
exten => ctx,n(doSkip),NoOp(${CALLERID(num)})
Thanks
zktech
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Re: [asterisk-users] Stripping characters from ${CALLERID(num)} ?
- Original Message -
> >> On Thu, 7 Jul 2011, Tim Nelson wrote:
> >>
> >>> On occasion, I have calls coming into an Asterisk 1.2.x system
> >>> where
> >>> the ${CALLERID(num)} includes '-'. Ex:
> >>>
> >>> 123-456-7890
> >>>
> >>> How can I strip the dashes from the number, leaving me with
> >>> '1234567890'?
>
> On Thu, 7 Jul 2011, Steve Edwards wrote:
>
> >> I would do this in an AGI written in C -- but that's just me...
>
> On Thu, 7 Jul 2011, Tim Nelson wrote:
>
> > Let's assume, "hypothetically", I'm looking for a simpler solution
> > directly in the dialplan or via a quick run of System(). :-)
>
> There is no such thing as 'a quick run of System()', relatively
> speaking.
>
> Even a simple system, echo, pipe, sed creates multiple processes.
> Process
> creation is among the most expensive operations (in terms of host
> resources) you can do.
Yes, I'm aware of that. By 'quick' I of course meant implementation effort, not
overall system resource usage. :)
>
> A single AGI is much more efficient. A few statements in dialplan are
> even
> more efficient in terms of host resources but can quickly spiral out
> of
> control as complexity rises.
I agree. In fact, I've noticed many of your posts are Asterisk 1.2.x and AGI/C
oriented which is very unique. Maybe lots of other people use AGI, but it
doesn't get much 'press' on the lists I guess. When I have some of that
mythical 'free time' I've been hearing so much about, it's on my list to learn
more about AGI. I know a little C, but would likely start out using PHPAGI as
it's more familiar to me. I know, not as efficient, but a stepping stone.
>
> I really like the 'black box' aspect of AGIs, especially when database
> access is added.
>
I'm currently running some database code in dialplan and its *NASTY*. If
anything, DB access would be the incentive for me to dive into AGI.
--Tim
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[asterisk-users] Installing Asterisk from repository works great without the need to install Dahdi on Host Node of Proxmox - But trying to install from source fails. Why?
Hi everyone, I just lunched a CentOS VM in Proxmox and used the Digium repository to install Asterisk using "yum install asterisk16"...and it works great. Runs and it seems to have installed ztdummy as well without the need to touch the host node. But when I try to compile Dahdi from source on the same VM to install Asterisk from source I get this: #@root/usr/src/dahdi/: *make all* make -C linux all make[1]: Entering directory `/usr/src/dahdi-linux-complete-2.4.1.2+2.4.1/linux' make -C drivers/dahdi/firmware firmware-loaders make[2]: Entering directory `/usr/src/dahdi-linux-complete-2.4.1.2+2.4.1/linux/drivers/dahdi/firmware' make[2]: Leaving directory `/usr/src/dahdi-linux-complete-2.4.1.2+2.4.1/linux/drivers/dahdi/firmware' *You do not appear to have the sources for the 2.6.32-4-pve kernel installed.* make[1]: *** [modules] Error 1 make[1]: Leaving directory `/usr/src/dahdi-linux-complete-2.4.1.2+2.4.1/linux' make: *** [all] Error 2 It seems that everyone is suggesting to install Dahdi on Host Node and then do modprobe ztdummy to get Dahdi running in VPS. Well, what is different between source install and repository install which doesn't need me to touch Host Node at all? I would rather not touch the Host Node at all and get a setup running just like Digium repository does. Any feedback is much appreciated. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] check_auth: username mismatch
> I've got a Polycom 501 that I just can't seem to get lines 2 and 3 to work > on. Line 1 works fine. Last time I had that issue, it resolved itself when i restarted Asterisk. Are you able to do that? Regards Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Stripping characters from ${CALLERID(num)} ?
On Thu, 7 Jul 2011, Tim Nelson wrote:
On occasion, I have calls coming into an Asterisk 1.2.x system where
the ${CALLERID(num)} includes '-'. Ex:
123-456-7890
How can I strip the dashes from the number, leaving me with
'1234567890'?
On Thu, 7 Jul 2011, Steve Edwards wrote:
I would do this in an AGI written in C -- but that's just me...
On Thu, 7 Jul 2011, Tim Nelson wrote:
Let's assume, "hypothetically", I'm looking for a simpler solution
directly in the dialplan or via a quick run of System(). :-)
There is no such thing as 'a quick run of System()', relatively speaking.
Even a simple system, echo, pipe, sed creates multiple processes. Process
creation is among the most expensive operations (in terms of host
resources) you can do.
A single AGI is much more efficient. A few statements in dialplan are even
more efficient in terms of host resources but can quickly spiral out of
control as complexity rises.
I really like the 'black box' aspect of AGIs, especially when database
access is added.
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Re: [asterisk-users] Polycom Blf / Directed Pickup
Oliver Your problem is you have not turned on "notifycid=yes" in sip.conf. Back on June 28 in another thread you said "With asterisk 1.6.1.18, I could make this work without setting notifycid=yes isn sip.conf." butyes that gets the monitored line to blink on an incoming call, but as you have discovered the phone will not do a directed pickup. This info is also available at http://www.voip-info.org/wiki/view/Asterisk+presence cheers gord On Wed, Jul 6, 2011 at 8:22 AM, Olivier wrote: > Using a Polycom 650 with 3.3.1, I could not have Directed Pickup working. > > More precisely, I configured the phone using and entries > as described in this thread. > Whenever a call comes in, BLF is blinking green. > Pressing the associated key generate generates a general Call Pickup (*8), > not a directed Call Pickup. > > Could you confirm this ? > > Regards > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Stripping characters from ${CALLERID(num)} ? [SOLVED]
- Original Message -
> On Thu, Jul 07, 2011 at 12:55:37PM -0500, Tim Nelson wrote:
> > - Original Message -
> > > On Thu, 7 Jul 2011, Tim Nelson wrote:
> > >
> > > > On occasion, I have calls coming into an Asterisk 1.2.x system
> > > > where
> > > > the
> > > > ${CALLERID(num)} includes '-'. Ex:
> > > >
> > > > 123-456-7890
> > > >
> > > > How can I strip the dashes from the number, leaving me with
> > > > '1234567890'?
> > >
> > > I would do this in an AGI written in C -- but that's just me...
> > >
> > >
> >
> > Let's assume, "hypothetically", I'm looking for a simpler solution
> > directly in the dialplan or via a quick run of System(). :-)
>
> I'm pretty sure that 1.2 has the FILTER dialplan function, so
>
> exten => Set(CALLERID(num)=${FILTER(0123456789|${CALLERID(num)})})
>
> should work.
>
This works perfectly, with the minor change of using ',' as the delimiter
instead of '|' with FILTER:
exten => _X.,n,Set(PROPERCID=${FILTER(1234567890,${CALLERID(num)})})
Thanks everyone for the help!
--Tim
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Re: [asterisk-users] SoftHangup on asterisk 1.8.2.3
On 7/7/2011 9:32 AM, Ishfaq Malik wrote:
I'm having the same issue on 1.8.3.2 (with a couple of patches)
exten => s,1,Set(CHAN=${SHELL(asterisk -rx "core show channels" | awk
'/^SIP\/vgw1-/ { print $1 }' | head -1)})
This turned out to be a PEBKAC error. A newline was attached to the
$CHAN variable.
adding | tr -d '\n' to the end of the command fixed it right up.
--
Jeremy Kister
http://jeremy.kister.net./
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Re: [asterisk-users] Stripping characters from ${CALLERID(num)} ?
On Thu, Jul 07, 2011 at 12:55:37PM -0500, Tim Nelson wrote:
> - Original Message -
> > On Thu, 7 Jul 2011, Tim Nelson wrote:
> >
> > > On occasion, I have calls coming into an Asterisk 1.2.x system where
> > > the
> > > ${CALLERID(num)} includes '-'. Ex:
> > >
> > > 123-456-7890
> > >
> > > How can I strip the dashes from the number, leaving me with
> > > '1234567890'?
> >
> > I would do this in an AGI written in C -- but that's just me...
> >
> >
>
> Let's assume, "hypothetically", I'm looking for a simpler solution directly
> in the dialplan or via a quick run of System(). :-)
I'm pretty sure that 1.2 has the FILTER dialplan function, so
exten => Set(CALLERID(num)=${FILTER(0123456789|${CALLERID(num)})})
should work.
--
Barry
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Re: [asterisk-users] Stripping characters from ${CALLERID(num)} ?
- Original Message -
> On Thu, 7 Jul 2011, Tim Nelson wrote:
>
> > On occasion, I have calls coming into an Asterisk 1.2.x system where
> > the
> > ${CALLERID(num)} includes '-'. Ex:
> >
> > 123-456-7890
> >
> > How can I strip the dashes from the number, leaving me with
> > '1234567890'?
>
> I would do this in an AGI written in C -- but that's just me...
>
>
Let's assume, "hypothetically", I'm looking for a simpler solution directly in
the dialplan or via a quick run of System(). :-)
--Tim
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Re: [asterisk-users] Stripping characters from ${CALLERID(num)} ?
On Thu, 7 Jul 2011, Tim Nelson wrote:
On occasion, I have calls coming into an Asterisk 1.2.x system where the
${CALLERID(num)} includes '-'. Ex:
123-456-7890
How can I strip the dashes from the number, leaving me with
'1234567890'?
I would do this in an AGI written in C -- but that's just me...
At the start of most of my client's dialplans I do something like:
exten = _!.,1, verbose(1,[${EXTEN}@${CONTEXT}])
exten = _!.,n, set(ANI=${EXTEN})
exten = _!.,n, agi(block-ani,--verbose)
The block-ani AGI does something like:
) Set the BLOCK channel variable to YES
) Read the ANI channel variable
) Strips leading '+'
) Strips leading '1'
) Strips anything non-numeric
) If the preceding steps changed anything, re-set the ANI channel variable
) Looks up the NPA to see if it should be blocked, return if true
) Looks up the NPA-NXX to see if it should be blocked, return if true
) Looks up the NPA-NXX- to see if it should be blocked, return if true
) Set the BLOCK channel variable to NO
) Return to the dialplan
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Re: [asterisk-users] Dropping Conference calls
Talk about getting lucky with a ! The first line you provided in
your log was enough to look-up related errors and find a similar one.
Although I have not encountered a Frame_Control(8), it's indicated as
'Congestion' by frame.h
AST_CONTROL_CONGESTION = 8, /*!< Congestion (circuits
busy) */
The one issue I found that appears similar (at least for the error/behavior)
was in relation to the admin's chan_dahdi.conf - the callprogress=yes option
to be specific. Perhaps posting your chan_dahdi.conf would be helpful in at
least verifying your settings are correct.
JT
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mark Rosedale
Sent: Wednesday, July 06, 2011 5:37 PM
To: jonathan.tho...@us.patersons.net
Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Dropping Conference calls
So I made the change you suggested. That still hasn't worked, but I did
manage to grab some logging from a dropped call.
[Jul 6 16:19:37] DEBUG[25950] channel.c: Got a FRAME_CONTROL (8) frame on
channel DAHDI/i1/18883203585-7e
[Jul 6 16:19:37] DEBUG[25950] res_rtp_asterisk.c: Setting the marker bit
due to a source update
[Jul 6 16:19:37] DEBUG[25950] chan_dahdi.c: Requested indication 20 on
channel DAHDI/i1/18883203585-7e
[Jul 6 16:19:37] DEBUG[25950] channel.c: Bridge stops bridging channels
SIP/7531-0077 and DAHDI/i1/18883203585-7e
[Jul 6 16:19:37] DEBUG[25950] cdr_mysql.c: Inserting a CDR record.
[Jul 6 16:19:37] DEBUG[25950] cdr_mysql.c: SQL command as follows: INSERT
INTO cdr
(`calldate`,`src`,`dst`,`dcontext`,`channel`,`dstchannel`,`lastapp`,`lastdat
a`,`duration`,`billsec`,`disposition`,`amaflags`,`accountcode`,`uniqueid`)
VALUES ('2011-07-06
15:58:57','7531','8883203585','from-sip','SIP/7531-0077','DAHDI/i1/18883
203585-7e','Dial','DAHDI/g1/18883203585','1240','1238','ANSWERED','3','\"Ada
m Witwer\"','1309982337.338')
[Jul 6 16:19:37] DEBUG[25950] channel.c: Hanging up channel
'DAHDI/i1/18883203585-7e'
[Jul 6 16:19:37] DEBUG[25950] chan_dahdi.c:
dahdi_hangup(DAHDI/i1/18883203585-7e)
[Jul 6 16:19:37] DEBUG[25950] chan_dahdi.c: Set option AUDIO MODE, value:
ON(1) on DAHDI/i1/18883203585-7e
[Jul 6 16:19:37] DEBUG[25950] sig_pri.c: sig_pri_hangup 1
[Jul 6 16:19:37] DEBUG[25950] sig_pri.c: Not yet hungup... Calling hangup
once with icause, and clearing call
[Jul 6 16:19:37] DEBUG[25950] chan_dahdi.c: Disabled echo cancellation on
channel 1
[Jul 6 16:19:37] DEBUG[25950] chan_dahdi.c: Set option TDD MODE, value:
OFF(0) on DAHDI/i1/18883203585-7e
[Jul 6 16:19:37] DEBUG[25950] chan_dahdi.c: Updated conferencing on 1, with
0 conference users
[Jul 6 16:19:37] DEBUG[25950] chan_dahdi.c: Set option AUDIO MODE, value:
OFF(0) on DAHDI/i1/18883203585-7e
[Jul 6 16:19:37] VERBOSE[25950] chan_dahdi.c: -- Hungup
'DAHDI/i1/18883203585-7e'
On Jul 1, 2011, at 2:38 PM, Jonathan Thomas wrote:
The exited non-zero is typical when a call has ended. What I would
recommend (easiest method) is for you to enter the CLI using: asterisk
-rvvv
The v's will provide more verbose logging, the 4 d's will place the core in
debug mode(4). Once in the CLI, pick a phone you will use as a test unit
and issue a
sip set debug peer XX (X=peer device id, such as 10001)
This will seriously increase the size of your logging - but should provide
you with a very thorough trace of the call as its in flight, including the
SIP dialog between the phone/server.
Perhaps you can enable the above, place a call that drops, then snip that
section of the full log and send it to the list for parsing. It's the best
way to nail down an issue like this.
JT
From: Mark Rosedale [mailto:mrosed...@oreilly.com]
Sent: Friday, July 01, 2011 2:17 PM
To: jonathan.tho...@us.patersons.net
Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Dropping Conference calls
So I didn't have sip debug set. So I don't have any SIP TIMER's in my log. I
have that set now.
I would be interested in the debut/logs if you have them.
I do have Spawn extension...exited non-zero on 'SIP/'
Here is the specifics
VERBOSE[10928] pbx.c: == Spawn extension (from-sip, 1***, 1) exited
non-zero on 'SIP/7XXX-09d7'
Not sure if that relates or not, but it is the only hit for the connection
between my sip client and the PRI going outbound right before the hangup.
On Jul 1, 2011, at 11:21 AM, Jonathan Thomas wrote:
The key item in my logs, which would preface the call dropping, was:
[2011-06-28 09:43:49] DEBUG[25563] chan_sip.c: ** SIP TIMER: Cancelling
retransmit of packet (reply received) Retransid #858
For instance - a call would be connected. SIP debug/core debug on. At the
14:30 mark I would begin tailing the full log. Once I saw the SIP TIMER
notice, it would be followed by a new INVITE (re-invite) SIP tr
[asterisk-users] Eyebeam crashes when dialing an invalid number...
Lately I have been getting many complains that Eyebeam crashes when you dial a number that does not exist. This happens in both R2 and ISDN PRI lines. The softphone stops working and has to be restarted. The response I got from tech support was: the actual issue is that asterisk should not be sending a 503 service unavailable when a particular softphone is not online. The soft phone stops because a 503 means that the server itself is unavailable. Does anyone have a workaround for this? Maybe a way to manipulate via dialplan so the softphone does not get the 503 message? -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Stripping characters from ${CALLERID(num)} ?
Greetings-
On occasion, I have calls coming into an Asterisk 1.2.x system where the
${CALLERID(num)} includes '-'. Ex:
123-456-7890
How can I strip the dashes from the number, leaving me with '1234567890'?
I've tried the following which does not appear to be working:
Dialplan:
exten => _X.,n,Set(PROPERCID=System(echo ${CALLERID(num)} | sed s/\-//g))
exten => _X.,n,NoOp(Fixed proper CID is ${PROPERCID}
Console Output:
-- Executing [11@cidmangletest:4] Set("SIP/w.x.y.z-b4d55ce8",
"PROPERCID=System(echo 123-456-7890 | sed s/\-//g)")
-- Executing [11@cidmangletest:5] NoOp("SIP/w.x.y.z-b4d55ce8",
"Fixed proper CID is System(echo 123-456-7890 | sed s/-//g)")
Obviously, I'm trying to throw the CID through sed via System() to strip the
dashes. Can anyone explain how to accomplish this? Or even better yet, how to
strip the dashes directly in the dialplan without the use of System()?
Thanks!
--Tim
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[asterisk-users] check_auth: username mismatch
Hi all, I've got a Polycom 501 that I just can't seem to get lines 2 and 3 to work on. Line 1 works fine. When my user tries to use line 2 or 3 to dial out, they get a fast busy signal and I get this error message on the console: === *CLI> [Jul 7 09:49:36] WARNING[26513]: chan_sip.c:12729 check_auth: username mismatch, have <0004F2127F60-1>, digest has <0004F2127F60-3> [Jul 7 09:49:36] NOTICE[26513]: chan_sip.c:20073 handle_request_invite: Failed to authenticate device "0004F2127F60-3" === All of my sip registrations are from a RT mysql database. The phone provisioning files are generated from this database. This is the only phone having the problem. I've rebooted the phone. The phone, and the database indicate that lines 2 and 3 are registered. 0004F2127F60-1 is the registration name for line 1 and 0004F2127F60-3 is for line 3. Any ideas on how to fix this? Would doing a factory reset and reprovisioning on the phone help? Or would that be just wheelspin? TIA, -- Take care and have fun, Mike Diehl. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] No pattern 407 from SIP provider iCall
Hi everyone, Occasionally (with no set pattern), I get *"SIP/2.0 407 Proxy Authentication Required" *from iCall when trying to termiate to their international gateways. I have tried direct IP termination as well as SIP register but both just fail with above message whenever they want. Specially in register mode where the user is registered and both userid and password are good and they have been good yesterday, today they fail and the next day they work. What could be the reason? Thanks, -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk 1.8X problem: no outbound callerid set in callfiles
Hi all,
Probably my message disappeared.
I updated from 1.4.42 to 1.8.4.4 and 1.8.5-rc1.
The problem is the same:
I generate a callfile with the option:
Callerid: test Callback Service <4711>
The callback is established correctly, but the variable ${CALLERID(num):} is
empty.
I don.t find: "test Callback Service <4711>"
on the DumpChan() or cli, or Master.csv.
Under the previous version (1.4.X) still has it all works fine.
Is this an undiscovered error?
In order to exclude errors, I have to compare the syntax in
sample.ccal.
Can your help please?
thanks.
-
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Re: [asterisk-users] SoftHangup on asterisk 1.8.2.3
I'm having the same issue on 1.8.3.2 (with a couple of patches)
Has anyone experienced this and know how to hangup a channel?
On Fri, 2011-02-04 at 17:25 -0500, Jeremy Kister wrote:
> I am trying to use SoftHangup in my dialplan, but it's either not
> working or I'm not using it correctly.
>
> when i'm on the console, i see:
> pbx1*CLI> core show channels
> Channel Location State Application(Data)
> SIP/vgw1-00a2 2156181505@inbound:1 Up AppDial((Outgoing Line))
> SIP/143-009f s@macro-SaferSIPDial Up Dial(SIP/99302156181505@vgw1,,
> 2 active channels
> 1 active call
> 194 calls processed
> pbx1*CLI>
>
>
> in my dialplan, i have:
> exten => s,1,Set(CHAN=${SHELL(asterisk -rx "core show channels" | awk
> '/^SIP\/vgw1-/ { print $1 }' | head -1)})
> exten => s,n,SoftHangup(${CHAN})
> exten => s,n,Wait(2)
>
>
>
> When I dial the extension to invoke the above dialplan code, the console
> shows:
> -- Executing [s@nineoneone:10] SoftHangup("SIP/111-00a3",
> "SIP/vgw1-00a2") in new stack
>
> but the SIP/vgw1-00a2 is still active. If I use 'channel request
> hangup SIP/vgw1-00a2', the call is dropped instantly.
>
> Am I using SoftHangup incorrectly?
>
>
--
Ishfaq Malik
Software Developer
PackNet Ltd
Office: 0161 660 3062
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Re: [asterisk-users] simple outbound call from asterisk to T1 card
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
> ADAK, INDRANIL (ATTSI)
> Sent: Thursday, July 07, 2011 8:58 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] simple outbound call from asterisk
> to T1 card
>
> Hello,
> I have a T1 card installed and is connected to PBX. I want
> to make outbound call to check if asterisk and t1 is working
> or not. The dhadi drivers are installed and the pri port is
> showing up and active. But calls are not going through. Can
> somebodu pdvise how to go about it ? I am sending you the
> call files and extensions.conf file details.
>
> Made a call file and
> channel: DAHDI/g0/9042021237@testing-context
> MaxRetries: 1
> RetryTime: 20
> WaitTime: 20
> Application: playback
> Data: demo-congrats
>
> *I also added the following to the extensions.conf file
> [globals] TRUNK_1=DAHDI/g0 [testing-context] exten =>
> _904XXX, 1, Dial(${TRUNK_1}/634,,D(86w904XXX))
>same => 2, playback(demo-congrats)
>same => 3, hangup()
> * I ALSO GET SAME WITH
> [globals]
> TRUNK_1=DAHDI/g0
> [testing-context]
> exten = s,1,Verbose(5,Dialing)
WRONG: channel: DAHDI/g0/9042021237@testing-context
RIGHT: channel: DAHDI/g0/9042021237
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[asterisk-users] simple outbound call from asterisk to T1 card
Hello,
I have a T1 card installed and is connected to PBX. I want to make
outbound call to check if asterisk and t1 is working or not. The dhadi
drivers are installed and the pri port is showing up and active. But
calls are not going through. Can somebodu pdvise how to go about it ? I
am sending you the call files and extensions.conf file details.
Made a call file and
channel: DAHDI/g0/9042021237@testing-context
MaxRetries: 1
RetryTime: 20
WaitTime: 20
Application: playback
Data: demo-congrats
*I also added the following to the extensions.conf file
[globals]
TRUNK_1=DAHDI/g0
[testing-context]
exten => _904XXX, 1, Dial(${TRUNK_1}/634,,D(86w904XXX))
same => 2, playback(demo-congrats)
same => 3, hangup()
* I ALSO GET SAME WITH
[globals]
TRUNK_1=DAHDI/g0
[testing-context]
exten = s,1,Verbose(5,Dialing)
same = n,Dial(DAHDI/g0/634,,D(86w9042021237))
Thanks
Indranil
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