Re: [asterisk-users] problems with hylafax + iaxmodem +asterisk1.8.5
G.Day! Thanks for the response! i've tryed to do this, but in /var/spool/hylafax/log/xferfaxlog I read this: 09/06/11 09:04 CALL00108 ttyIAXfax +39.06.456789 0 0 0:00:09 0:00:09 Failure to receive silence (synchronization failure). 06654321 NONE::s what is it?! -- From: Larry Moore lmo...@starwon.com.au Sent: Monday, September 05, 2011 10:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] problems with hylafax + iaxmodem +asterisk1.8.5 On 5/09/2011 10:05 PM, Alessio wrote: someone can help me to solve this problem? thanks -- From: Alessio ales...@asistar.it Sent: Friday, September 02, 2011 5:10 PM To: Lee Howard fax...@howardsilvan.com Cc: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] problems with hylafax + iaxmodem +asterisk1.8.5 1: from the phone i called the fax-server 2: from external fax i tried to send a fax to fax-server the results: _ G'Day Alessio, I replied to your original post suggesting you set up two IAX modems and get successful transmission working between them. I suspect you want to use T.38 with IAX modem, I don't believe the IAX2 channel supports T.38 hence I would suggest you remove the t38pt_udptl lines from your iax.conf files to avoid confusion. I am assuming you are receiving your incoming facsimile using SIP, if so I would suggest you have only one reference to t38pt_udptl in that peers configuration and set it to no. Depending on whether the peer is dedicated to receiving facsimiles I would suggest you also include in your peer's configuration faxdetect=no otherwise if this is an Audio/FAX line I would suggest you set it to faxdetect=cng. Once you have this working but really want to use T.38 then you will need to apply the T.38 Gateway patch to your 1.8.5.0 build, see https://issues.asterisk.org/view.php?id=13405 . Changes you will need to make to your SIP peer is to set t38pt_udptl=yes and in your dial plan before the Dial() enable the gateway with Set(FAXOPT(t38gateway)=yes). Larry. -- _ -- 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
[asterisk-users] ISDN2 PCIe Card for Asterisk
Hi, I'm looking for a PCIe card with 1 ISDN2 connection which works with Asterisk Could anybody give me an advise which card I can use? Regards, Arjan Kroon Mobillion. -- _ -- 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] ISDN2 PCIe Card for Asterisk
what do you mean exactly?! One what?! What do you plan to accomplish?! Do you mean a 1 Port ISDN BRI Board?! Difficult to find, and thus boards are really expensive, not under 400.- € inkluding DSP Processors. I advise you taking Gentoo Linux, getting asterisk on it and put a single Port HFC-S PCI (not PCIe) Board in your CPU. If you need something really professional, for Serverside, I advise you sangoma. Tamer Am 06.09.2011 09:08, schrieb Arjan Kroon | Mobillion: Hi, I'm looking for a PCIe card with 1 ISDN2 connection which works with Asterisk Could anybody give me an advise which card I can use? Regards, Arjan Kroon Mobillion. -- _ -- 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] ISDN2 PCIe Card for Asterisk
Hi Tamar, Yes, I mean 1 Port ISDN BRI PCIe board. We need an PCIe board, because the board don't provide PCI slots, only PCIe slots. It doesn't matter which distribution we use. But I will look at Sangoma. Best Regards, Arjan Mobillion BV -Oorspronkelijk bericht- Van: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Namens Tamer Higazi Verzonden: 06-09-2011 10:39 Aan: Asterisk Users Mailing List - Non-Commercial Discussion Onderwerp: Re: [asterisk-users] ISDN2 PCIe Card for Asterisk what do you mean exactly?! One what?! What do you plan to accomplish?! Do you mean a 1 Port ISDN BRI Board?! Difficult to find, and thus boards are really expensive, not under 400.- € inkluding DSP Processors. I advise you taking Gentoo Linux, getting asterisk on it and put a single Port HFC-S PCI (not PCIe) Board in your CPU. If you need something really professional, for Serverside, I advise you sangoma. Tamer Am 06.09.2011 09:08, schrieb Arjan Kroon | Mobillion: Hi, I'm looking for a PCIe card with 1 ISDN2 connection which works with Asterisk Could anybody give me an advise which card I can use? Regards, Arjan Kroon Mobillion. -- _ -- 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] ISDN2 PCIe Card for Asterisk
On 09/06/2011 09:08 AM, Arjan Kroon | Mobillion wrote: Hi, I'm looking for a PCIe card with 1 ISDN2 connection which works with Asterisk Could anybody give me an advise which card I can use? Both Sangoma and Digium have PCIe ISDN cards although a single BRI port might be a bit of a challenge: http://sangoma.com/products/hardware_products/digital_voice_and_data_networking/a500.html http://sangoma.com/products/hardware_products/digital_analog_hybrids/b700_flex_bri.html http://www.digium.com/en/products/hybrid/h8.php You could also find an Intel (formerly Eicon) Diva Server card and use it with chan_capi (must be a Server card, regular Diva cards don't work). I use that in one Asterisk box and it's very reliable. Groet, Patrick -- _ -- 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] PARAMETERS FOR DID'S FROM TELCOM AND ASTERISK TO SYNC
Hi list, I have 20channels PRI line from Airtel. and I have 30channels digium PRI card. and I am using asterisk1.4 with goautodial. I need to configure DID's for every extension with sip. which Parameters need to add in asterisk and which parameters enable from Airtel PRI line for this DID's. can you help me . Best Regards, Mahesh Katta *BUZZ**WORKS* Business Services Private Limited BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI 201, Crystal Tower, 75 Gundavali Cross Lane, Andheri Kurla Road, Andheri (E) Mumbai 400069 GSM +91.97029.70779 | Phone +91.22.4229.2634 | Fax +91.22.4229.2634 Web http://www.buzzworks.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
Re: [asterisk-users] Distributed device state / presence info??
On 09/05/2011 03:05 AM, Hans Witvliet wrote: On Fri, 2011-09-02 at 11:33 -0500, Kevin P. Fleming wrote: On 09/01/2011 04:39 PM, Hans Witvliet wrote: From the asterisk-bible and the wiki's i learned that it is possible to let asterisk do some of the presense-info by means of the jabber.conf file and a seperate xmpp-server. You are misunderstanding a bit; Asterisk can use an XMPP server and PubSub to *distribute* presence information among a cluster of Asterisk servers. This information is not intended to be directly sent to XMPP clients. What i assume (please correct me if i am wrong) is that when a client registers/deregisters, asterisk will update the presence info towards the XMPP-server. Correct? Yes, in order to let other Asterisk servers in the cluster know about it. But otoh, what people would like to see is who is on line. And not only on the asterisk-server that they are connected to, but also from other possible asterisk servers. And furthermore, each registered user might want to set their presencse-status to either free/busy/away/what-ever. Asterisk does not support 'user' presence; it supports device and extension presence. In some applications these can be used interchangeably, but in others they don't match up very well. Ok, so that should mean that the presence-status is controlled by the fect wether a sip-user is registered or not? If so, i'm still making a mistake somewhere: No, Asterisk sends extension state information, not presence. The informations sent to the SIP endpoints is based on whether the monitored endpoint is considered 'busy' or not. It may also be based on whether it is registered or not, but I don't remember for sure. I tried to simplify my configuration: Just one asterisk machine, And two users (my and myself) with two linphones, one from a XP-machine and the other from a SuSE-machine. They can both register and call each other, and in linphone presence is enabled. However if i manually add on each side the corresponding info (name + sip-address) in the linphone app, they always remain grey / away Even while in the middle of an connection to each other. Have you created hints in the dialplan to tell Asterisk what device(s) it should monitor in order to provide the extension state for the URIs you are asking the phones to subscribe to? -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 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 -- 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] Beggining asterisk
On 04/09/11 02:51 PM, Tamer Higazi wrote: the 3rd edition is available, but that book covers every thing to run the asterisk PBX. You can read the 3rd edition online at http://ofps.oreilly.com/titles/9780596517342/ HTH! Leif. -- Leif Madsen http://www.oreilly.com/catalog/asterisk -- _ -- 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] PARAMETERS FOR DID'S FROM TELCOM AND ASTERISK TO SYNC
On Tuesday 06 September 2011, mahesh katta wrote:
Hi list,
I have 20channels PRI line from Airtel. and I have 30channels digium PRI
card. and I am using asterisk1.4 with goautodial.
I need to configure DID's for every extension with sip.
which Parameters need to add in asterisk and which parameters enable from
Airtel PRI line for this DID's. can you help me .
When an incoming call arrives from the ISDN line, look in ${EXTEN} -- this
will contain some representation of the number. It may be missing the STD
code, or just the initial 0.
All* you need to do is implement a rule to translate the incoming ${EXTEN} on
an incoming call, to an extension number, and implement this in your dialplan
in the conjtext in which calls arrive from the PSTN. For example, if you
always want the internal number to be 2 followed by the last 2 digits of
the external number, you could use this minimal example:
[from-pstn]
exten = _X.,1,Set(LAST_TWO=${EXTEN:-2})
exten = _X.,n,Dial(SIP/2${LAST_TWO})
exten = _X.,n,Hangup()
[*] OK, that's not *quite* all. When you make an outgoing call, you need to
set the caller ID to match the external number the person needs to dial to
call you back:
[outgoing]
exten = _X.,1,Set(LAST_TWO=${EXTEN:-2})
exten = _X.,n,Set(CALLERPRES()=allowed)
exten = _X.,n,Set(CallerID(num)=${PREFIX}${LAST_TWO})
exten = _X.,n,Dial(${ISDN}/${EXTEN})
exten = _X.,n,Hangup()
${ISDN} and ${PREFIX} need to be defined in the globals section:
[globals]
ISDN=DAHDI/g1
PREFIX=013322689
; this assumes numbers start (01332) 2689xx
--
AJS
Answers come *after* questions.
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Re: [asterisk-users] Beggining asterisk
2011/9/6 Leif Madsen leif.mad...@asteriskdocs.org On 04/09/11 02:51 PM, Tamer Higazi wrote: the 3rd edition is available, but that book covers every thing to run the asterisk PBX. You can read the 3rd edition online at http://ofps.oreilly.com/** titles/9780596517342/ http://ofps.oreilly.com/titles/9780596517342/ HTH! Leif. -- Leif Madsen http://www.oreilly.com/**catalog/asteriskhttp://www.oreilly.com/catalog/asterisk -- __**__**_ -- 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-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users Thanks for all the responses !. I will try with ubuntu bundleded packages first. I will post my results. Esteban -- Esteban L. Cacavelos de Amoriza Cel: 0981 220 429 -- _ -- 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] PARAMETERS FOR DID'S FROM TELCOM AND ASTERISK TO SYNC
Thanks For reply A.J sir.
no result with this sir
Best Regards,
Mahesh Katta
*BUZZ**WORKS* Business Services Private Limited
BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI
201, Crystal Tower, 75 Gundavali Cross Lane, Andheri Kurla Road, Andheri (E)
Mumbai 400069
GSM +91.97029.70779 | Phone +91.22.4229.2634 | Fax +91.22.4229.2634
Web http://www.buzzworks.com
On Tue, Sep 6, 2011 at 6:49 PM, A J Stiles asterisk_l...@earthshod.co.ukwrote:
On Tuesday 06 September 2011, mahesh katta wrote:
Hi list,
I have 20channels PRI line from Airtel. and I have 30channels digium PRI
card. and I am using asterisk1.4 with goautodial.
I need to configure DID's for every extension with sip.
which Parameters need to add in asterisk and which parameters enable from
Airtel PRI line for this DID's. can you help me .
When an incoming call arrives from the ISDN line, look in ${EXTEN} -- this
will contain some representation of the number. It may be missing the STD
code, or just the initial 0.
All* you need to do is implement a rule to translate the incoming ${EXTEN}
on
an incoming call, to an extension number, and implement this in your
dialplan
in the conjtext in which calls arrive from the PSTN. For example, if you
always want the internal number to be 2 followed by the last 2 digits
of
the external number, you could use this minimal example:
[from-pstn]
exten = _X.,1,Set(LAST_TWO=${EXTEN:-2})
exten = _X.,n,Dial(SIP/2${LAST_TWO})
exten = _X.,n,Hangup()
[*] OK, that's not *quite* all. When you make an outgoing call, you need
to
set the caller ID to match the external number the person needs to dial
to
call you back:
[outgoing]
exten = _X.,1,Set(LAST_TWO=${EXTEN:-2})
exten = _X.,n,Set(CALLERPRES()=allowed)
exten = _X.,n,Set(CallerID(num)=${PREFIX}${LAST_TWO})
exten = _X.,n,Dial(${ISDN}/${EXTEN})
exten = _X.,n,Hangup()
${ISDN} and ${PREFIX} need to be defined in the globals section:
[globals]
ISDN=DAHDI/g1
PREFIX=013322689
; this assumes numbers start (01332) 2689xx
--
AJS
Answers come *after* questions.
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Re: [asterisk-users] PARAMETERS FOR DID'S FROM TELCOM AND ASTERISK TO SYNC
On Tuesday 06 September 2011, mahesh katta wrote:
Thanks For reply A.J sir.
no result with this sir
OK. no result isn't very helpful.
I'm presuming you've got internal calls between SIP extensions working
correctly, and something happens (even if it's not exactly what you want)
when you call an inbound number. If not, get at least this much working first.
I'm also presuming that you know a few things about dialplans, and can make
appropriate substitutions in your head.
In your chan_dahdi.conf, what is the name of the context associated with the
span to which the ISDN is connected? (I assumed in my example that the
relevant context was called from-pstn.) Make a note of this.
Now make a copy of your existing extensions.conf, then open the original in a
text editor. In the context where calls from the ISDN line arrive, create the
following extension:
exten = .X_,1,NoOp(Incoming call for '${EXTEN} from ${CALLERID(num)}')
exten = .X_,n,Hangup()
Open an Asterisk console with maximum verbosity, run dialplan reload, call
an inbound number from a mobile phone and you should get a message something
like
Incoming call for '01332268901' from '07892101232'
with some or all of the number you dialled and your mobile number, just before
the call disconnects. (If not, then something else is wrong.)
Now you know how much of the dialled number the phone company are sending you
(complete number / number without leading 0 / local number without STD code).
Next, determine what the rule is going to be to link the dialled number in
${EXTEN} to an individual internal extension (in my example, I assumed that
the rule was: 2 followed by the last 2 digits of the external number)
and devise a dialplan expression that will set a variable to that. (If you
don't know how to do this, please ask, giving some examples I can work from.)
Now, between the NoOp() and the Hangup() lines in our minimal context as
above, insert lines to set a variable from your expression and another NoOp()
to display its value.
Open an Asterisk console with maximum verbosity, run dialplan reload again
to make sure you are using the correct dialplan, call an inbound number from a
mobile phone, and this time you should get another message with the internal
extension number you want to call. Check it with as many numbers as you can
afford to waste credit on.
When you're sure that you have your formula working, add a Dial() statement
which will dial that number. If you want, remove the NoOp() statements;
remember to make the statement which is now first, priority 1 instead of n.
Again, run dialplan reload before you make an inbound call, just to ensure
you are using the latest edited dialplan.
That should sort the incoming side. On the outgoing side, you need to write a
dialplan expression which will set CALLERID(num) to whatever external number
is associated with an internal extension (which will be in ${CALLERID(num)}
-- sorry, I made a mistake and put ${EXTEN} in my first example). basically
the inverse of the first rule. Then put this into your context which is used
for outgoing calls.
If any of this is going over your head, please ask for clarification. (My
preference is generally to overestimate someone's abilities and rely on them
asking questions if lost, rather than sound like I am patronising them.)
--
AJS
Answers come *after* questions.
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Re: [asterisk-users] ISDN2 PCIe Card for Asterisk
On Tue, Sep 06, 2011 at 01:05:02PM +0200, Patrick Lists wrote: Both Sangoma and Digium have PCIe ISDN cards although a single BRI port might be a bit of a challenge: [snip] You could also find an Intel (formerly Eicon) Diva Server card and use it with chan_capi (must be a Server card, regular Diva cards don't work). I use that in one Asterisk box and it's very reliable. I personally decided against using PCI(e) ISDN BRI cards. Eicon and various AVM cards all had problems with echo cancelling. I'd suggest using external SIP gateways like Vegastream or SmartNodes. Easy to configure (once you figure out how with a SmartNode), good quality and much easier in failover (though might be a single point of failure by itself). -- Daniel Tryba -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] pick up code
Hello list
i want to use pickup with sip and astersik 1.4
i configured all the inbound calls in 1 sip phone 224 and want to pickup the
calls using 222 SIP
Could you please see the code below and tell me what is wrong
NB when i make *8+ok i can pickup the call but i want to specify the number.
extenssion.conf
[agents]
exten = _2XX,1,Dial(SIP/${EXTEN})
exten = _*8XXX,1,PickupChan(SIP/${EXTEN:2})
sip.conf
[general]
context=agents
allowguest=yes
allowoverlap=no
allowtransfer=yes
allow=alaw
allow=ulaw
allow=gsm
allow=ilbc
[222]
type=friend
context=agents
host=dynamic
dtmfmode=auto
disallow=all
allow=alaw
allow=ulaw
qualify=yes
callgroup=1
pickupgroup=1
[224]
type=friend
context=agents
host=dynamic
dtmfmode=auto
disallow=all
allow=alaw
allow=ulaw
qualify=yes
callgroup=1
pickupgroup=1
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Re: [asterisk-users] How does AMI work with events ?
On Mon, Sep 05, 2011 at 02:41:50PM +0200, Jonas Kellens wrote: I read some information and examples on the net, but they all show how you login to the AMI, give an action and receive a response. The end. I guess you just re-run the script every time you want the action to be executed. How then does this work when you use events ? If I want to use PeerStatus to monitor the state of a SIP peer, how can I run a script on change of peer status ? [snip] That's it. I'm logged in and I have been subscribed to receive peer status changes. Where do these peer changes appear ? I really don't get that. This is fairly basic stuff: 1-connect to AMI 2-authenticate 3-subscribe to events 4-listen for events 5-check if wanted event and do something with it, else 6-go back to listening So in your script you are connected, just start listening (fgets()) and check if the appropriate event arrived (keep reading lines till first empty line). Parse the event and do something (mail in your case) when required. Examples of PeerStats Events: http://www.voip-info.org/wiki/view/asterisk+manager+events#PeerStatusEvent Creating a prototype daemon in PHP should be easy, but I experienced weird hangups in PHP with longlived sockets. Personally I monitor SIP/IAX peers by parsing: asterisk -nrx sip show peers with Nagios/NRPE scripts. -- Daniel Tryba -- _ -- 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] pick up code
On Tue, Sep 06, 2011 at 04:43:39PM +, salaheddine elharit wrote:
[asterisk 1.4]
[agents]
exten = _2XX,1,Dial(SIP/${EXTEN})
exten = _*8XXX,1,PickupChan(SIP/${EXTEN:2})
SIP/222 is not a channel but an extension. See:
http://www.voip-info.org/wiki/view/Asterisk+cmd+Pickup
--
Daniel Tryba
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Re: [asterisk-users] pick up code
ok thanks for you response
how can i do in order to fix this issue
regards
2011/9/6 Daniel Tryba dan...@tryba.nl
On Tue, Sep 06, 2011 at 04:43:39PM +, salaheddine elharit wrote:
[asterisk 1.4]
[agents]
exten = _2XX,1,Dial(SIP/${EXTEN})
exten = _*8XXX,1,PickupChan(SIP/${EXTEN:2})
SIP/222 is not a channel but an extension. See:
http://www.voip-info.org/wiki/view/Asterisk+cmd+Pickup
--
Daniel Tryba
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Re: [asterisk-users] Variables error in 1.8.6.0.
Hello Leandro,
Can you tell me a short example about how can i use what you gave me for
instance suppose i want to use { txjitter, DBL, { .d8 =
stats.txjitter, }, }, how can i set it in CDR variable like mine:
exten = h,n,set(CDR(ljitt)=${CHANNEL(rtpqos,audio,local_jitter)})
Thank you.
On Mon, Sep 5, 2011 at 10:58 PM, Leandro Dardini ldard...@gmail.com wrote:
2011/9/5 Catalin S. jonsonpla...@gmail.com
Hello,
I have a problem with some variables in 1.8.6.0. I set on extension the
following lines:
exten = h, n, Set (CDR (LLP) = $ {CHANNEL (rtpqos, audio,
local_lostpackets)}) ; lost packets by local end **
exten = h, n, Set (CDR (PCR) = $ {CHANNEL (rtpqos, audio,
remote_lostpackets)}) ; lost packets by remote end
exten = h, n, Set (CDR (ljitt) = $ {CHANNEL (rtpqos, audio,
local_jitter)}) ; the Same for jitter
Theoretically this should throw these variables in a table in MySQL but
these values cannot be readed. I think it's a different syntax in
1.8.
I gave this error:
- Executing [h @ macro-special1: 11] Set (SIP/1010-0002, CDR
(LLP) =) in new stack
[September 5 22:39:33] WARNING [14432]: SIP / dialplan_functions.c: 221
sip_acf_channel_read: Unrecognized argument 'rtpqos, audio,
remote_lostpackets' to CHANNEL
[September 5 22:39:33] WARNING [14432]: func_channel.c: 393
func_channel_read: Unknown or unavailable item Requested 'rtpqos, audio,
remote_lostpackets'
- Executing [h @ macro-special1: 12] Set (SIP/1010-0002, CDR
(PCR) =) in new stack
[September 5 22:39:33] WARNING [14432]: SIP / dialplan_functions.c: 221
sip_acf_channel_read: Unrecognized argument 'rtpqos, audio, local_jitter' to
CHANNEL
[September 5 22:39:33] WARNING [14432]: func_channel.c: 393
func_channel_read: Unknown or unavailable item Requested 'rtpqos, audio,
local_jitter'
- Executing [h @ macro-special1: 13] Set (SIP/1010-0002, CDR
(ljitt) =) in new stack
[September 5 22:39:33] WARNING [14432]: SIP / dialplan_functions.c: 221
sip_acf_channel_read: Unrecognized argument 'rtpqos, audio, remote_jitter'
to CHANNEL
[September 5 22:39:33] WARNING [14432]: func_channel.c: 393
func_channel_read: Unknown or unavailable item Requested 'rtpqos, audio,
remote_jitter'
Any idea how I can fix?
Best regards,
Jonson.
--
It is really simple, a patch of few months ago renamed the vars, but
forget to update the documentation. You have to use the source for finding
the new variable names. I paste here the part of the code for your easy
viewing...
{ txcount, INT, { .i4 =
stats.txcount, }, },
{ rxcount, INT, { .i4 =
stats.rxcount, }, },
{ txjitter, DBL, { .d8 =
stats.txjitter, }, },
{ rxjitter, DBL, { .d8 =
stats.rxjitter, }, },
{ remote_maxjitter, DBL, { .d8 =
stats.remote_maxjitter, }, },
{ remote_minjitter, DBL, { .d8 =
stats.remote_minjitter, }, },
{ remote_normdevjitter, DBL, { .d8 =
stats.remote_normdevjitter, }, },
{ remote_stdevjitter,DBL, { .d8 =
stats.remote_stdevjitter, }, },
{ local_maxjitter, DBL, { .d8 =
stats.local_maxjitter, }, },
{ local_minjitter, DBL, { .d8 =
stats.local_minjitter, }, },
{ local_normdevjitter, DBL, { .d8 =
stats.local_normdevjitter, }, },
{ local_stdevjitter, DBL, { .d8 =
stats.local_stdevjitter, }, },
{ txploss, INT, { .i4 =
stats.txploss, }, },
{ rxploss, INT, { .i4 =
stats.rxploss, }, },
{ remote_maxrxploss, DBL, { .d8 =
stats.remote_maxrxploss, }, },
{ remote_minrxploss, DBL, { .d8 =
stats.remote_minrxploss, }, },
{ remote_normdevrxploss, DBL, { .d8 =
stats.remote_normdevrxploss, }, },
{ remote_stdevrxploss, DBL, { .d8 =
stats.remote_stdevrxploss, }, },
{ local_maxrxploss, DBL, { .d8 =
stats.local_maxrxploss, }, },
{ local_minrxploss, DBL, { .d8 =
stats.local_minrxploss, }, },
{ local_normdevrxploss, DBL, { .d8 =
stats.local_normdevrxploss, }, },
{ local_stdevrxploss,DBL, { .d8 =
stats.local_stdevrxploss, }, },
{ rtt, DBL, { .d8 =
stats.rtt, }, },
{ maxrtt,DBL, { .d8 =
stats.maxrtt, }, },
[asterisk-users] trying to build 1.8.6.0 on CentOS 6, problems with ptlib
I'm having annoying errors trying to get configure working. tar xvzf /usr/local/src/asterisk-1.8.6.0.tar.gz cd asterisk-1.8.6.0 ./configure I get complaints related to pwlib / ptlib... checking for openr2_chan_new in -lopenr2... no checking /root/pwlib/include/ptlib.h usability... no checking /root/pwlib/include/ptlib.h presence... no checking for /root/pwlib/include/ptlib.h... no checking /usr/local/include/ptlib.h usability... no checking /usr/local/include/ptlib.h presence... no checking for /usr/local/include/ptlib.h... no checking /usr/include/ptlib.h usability... yes checking /usr/include/ptlib.h presence... yes checking for /usr/include/ptlib.h... yes checking for ptlib-config... /usr/bin/ptlib-config ./configure: line 24978: 2*1+6*100+5 /usr/lib64/ /usr/lib/: division by 0 (error token is /lib64/ /usr/lib/) There seems to also be a problem with CentOS 6 in general that I have not found a package that actually provides /usr/bin/ptlib-config. I copied that binary over from a CentOS 5 install to see if I could get my original error to clear. Here's THAT error... checking for openr2_chan_new in -lopenr2... no checking /root/pwlib/include/ptlib.h usability... no checking /root/pwlib/include/ptlib.h presence... no checking for /root/pwlib/include/ptlib.h... no checking /usr/local/include/ptlib.h usability... no checking /usr/local/include/ptlib.h presence... no checking for /usr/local/include/ptlib.h... no checking /usr/include/ptlib.h usability... yes checking /usr/include/ptlib.h presence... yes checking for /usr/include/ptlib.h... yes checking for ptlib-config... no ./configure: line 24906: --ldflags: command not found Cannot find ptlib-config - please install and try again And then I searched around the config trying to figure out how not to use ptlib at all, and ultimately I tried just doing... rpm -e ptlib ptlib-devel (and the other packages) And this got my asterisk configure to complete. But really, I'm wondering if other people have run into the same problem, or if there is a nifty configure argument that would keep me from needing to uninstall the library. -- _ -- 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] trying to build 1.8.6.0 on CentOS 6, problems with ptlib
On 09/06/2011 03:08 PM, David Backeberg wrote: There seems to also be a problem with CentOS 6 in general that I have not found a package that actually provides /usr/bin/ptlib-config. I copied that binary over from a CentOS 5 install to see if I could get my original error to clear. Here's THAT error... checking for openr2_chan_new in -lopenr2... no checking /root/pwlib/include/ptlib.h usability... no checking /root/pwlib/include/ptlib.h presence... no checking for /root/pwlib/include/ptlib.h... no checking /usr/local/include/ptlib.h usability... no checking /usr/local/include/ptlib.h presence... no checking for /usr/local/include/ptlib.h... no checking /usr/include/ptlib.h usability... yes checking /usr/include/ptlib.h presence... yes checking for /usr/include/ptlib.h... yes checking for ptlib-config... no ./configure: line 24906: --ldflags: command not found Cannot find ptlib-config - please install and try again This is a bug in the configure script, but in the meantime, you should be able to use --without-pwlib to avoid it, as long as you aren't trying to build chan_h323. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 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 -- 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] Set(CHANNEL(musicclass)=
On 02/09/11 11:27 PM, Joseph wrote: In asterisk 1.4 I had: exten = s,n,Answer() exten = s,n,SetMusicOnHold(default) But in 1.6 1.8 I think don't need to use: SetMusicOnHold(default) (beside it is deprecated) as it is default. In 1.6 and UP I think it is: Set(CHANNEL(musicclass)= Can somebody explain what do they mean by CHANNEL? CHANNEL() is a dialplan function. You're setting parameters for the current channel by using that function. So instead of using a dialplan application like you were before, you use the CHANNEL() function. exten = s,1,NoOp() same = n,Set(CHANNEL(musicclass)=default) I could use just: exten = s,n,MusicOnHold() There is a lot of documentation on www.voip-info.org but sometimes it is not clear which asterisk version it applies to :-/ (Another good reason to be reading the documentation on https://wiki.asterisk.org/wiki instead :)) -- Leif Madsen http://www.oreilly.com/catalog/asterisk -- _ -- 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] trying to build 1.8.6.0 on CentOS 6, problems with ptlib
First, have you tried ./configure --help? Next try the --with-pwlib parameter Somewhere in the list, make sure your YUM paths are happy. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Backeberg Sent: Tuesday, September 06, 2011 3:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] trying to build 1.8.6.0 on CentOS 6, problems with ptlib I'm having annoying errors trying to get configure working. tar xvzf /usr/local/src/asterisk-1.8.6.0.tar.gz cd asterisk-1.8.6.0 ./configure I get complaints related to pwlib / ptlib... checking for openr2_chan_new in -lopenr2... no checking /root/pwlib/include/ptlib.h usability... no checking /root/pwlib/include/ptlib.h presence... no checking for /root/pwlib/include/ptlib.h... no checking /usr/local/include/ptlib.h usability... no checking /usr/local/include/ptlib.h presence... no checking for /usr/local/include/ptlib.h... no checking /usr/include/ptlib.h usability... yes checking /usr/include/ptlib.h presence... yes checking for /usr/include/ptlib.h... yes checking for ptlib-config... /usr/bin/ptlib-config ./configure: line 24978: 2*1+6*100+5 /usr/lib64/ /usr/lib/: division by 0 (error token is /lib64/ /usr/lib/) There seems to also be a problem with CentOS 6 in general that I have not found a package that actually provides /usr/bin/ptlib-config. I copied that binary over from a CentOS 5 install to see if I could get my original error to clear. Here's THAT error... checking for openr2_chan_new in -lopenr2... no checking /root/pwlib/include/ptlib.h usability... no checking /root/pwlib/include/ptlib.h presence... no checking for /root/pwlib/include/ptlib.h... no checking /usr/local/include/ptlib.h usability... no checking /usr/local/include/ptlib.h presence... no checking for /usr/local/include/ptlib.h... no checking /usr/include/ptlib.h usability... yes checking /usr/include/ptlib.h presence... yes checking for /usr/include/ptlib.h... yes checking for ptlib-config... no ./configure: line 24906: --ldflags: command not found Cannot find ptlib-config - please install and try again And then I searched around the config trying to figure out how not to use ptlib at all, and ultimately I tried just doing... rpm -e ptlib ptlib-devel (and the other packages) And this got my asterisk configure to complete. But really, I'm wondering if other people have run into the same problem, or if there is a nifty configure argument that would keep me from needing to uninstall the library. -- _ -- 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] trying to build 1.8.6.0 on CentOS 6, problems with ptlib
On Tue, Sep 6, 2011 at 4:28 PM, Kevin P. Fleming kpflem...@digium.com wrote: This is a bug in the configure script, but in the meantime, you should be able to use --without-pwlib to avoid it, as long as you aren't trying to build chan_h323. Thanks much. I was trying ./configure --disable-chan_ooh323 and that was not making a difference. -- _ -- 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] trying to build 1.8.6.0 on CentOS 6, problems with ptlib
On 09/06/2011 04:09 PM, David Backeberg wrote: On Tue, Sep 6, 2011 at 4:28 PM, Kevin P. Flemingkpflem...@digium.com wrote: This is a bug in the configure script, but in the meantime, you should be able to use --without-pwlib to avoid it, as long as you aren't trying to build chan_h323. Thanks much. I was trying ./configure --disable-chan_ooh323 and that was not making a difference. It won't, for two reasons: Asterisk modules can't be selected/deselected via the configure script (menuselect is used for that), and chan_ooh323 doesn't use pwlib/openh323, chan_h323 does. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 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 -- 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] Beggining asterisk
2011/9/6 Esteban Cacavelos estebancacave...@gmail.com
2011/9/6 Leif Madsen leif.mad...@asteriskdocs.org
On 04/09/11 02:51 PM, Tamer Higazi wrote:
the 3rd edition is available, but that book covers every thing to run
the asterisk PBX.
You can read the 3rd edition online at http://ofps.oreilly.com/**
titles/9780596517342/ http://ofps.oreilly.com/titles/9780596517342/
HTH!
Leif.
--
Leif Madsen
http://www.oreilly.com/**catalog/asteriskhttp://www.oreilly.com/catalog/asterisk
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Thanks for all the responses !. I will try with ubuntu bundleded packages
first.
I will post my results.
Esteban
--
Esteban L. Cacavelos de Amoriza
Cel: 0981 220 429
finally i decided to install from source because the documentation suggest
that.
I've installed successfully asterisk+dahdi+libpri. I tested a basic SIP
configuration and there were no problems.
Now i have problems with pstn termination and origination. I have one fxo
module from witch i want to make and receive calls. Can I do that ?. I'll
post my configuration files.
I want to make calls from my android phone (where i have a SIP client) and
recieve calls from my analog line through my androi.
My country code is 595, city code 21, number , xxx xxx
chan_dahdi.conf
[channels]
;
; To apply other options to these channels, put them before channel.
;
signalling=fxs_ks ; in Asterisk, FXO channels use FXS signaling
; (and yes, FXS channels use FXO signaling)
context=from-pstn
channel = 1 ; apply all the previously defined settings to this
channel
extensions.conf
[LocalSets]
exten = 100,1,Dial(SIP/android-esteban) ; Replace 0001 with your
device name
exten = 101,1,Dial(SIP/recepcion) ; Replace 0002 with your device
name
exten = 200,1,Answer()
same = n,Playback(hello-world)
same = n,Hangup()
; TERMINATION
[from-voip-network]
exten = _X.,1,Verbose(2, Call from VoIP network to ${EXTEN})
same = n,Dial(DAHDI/g0/${EXTEN})
ORIGINATION
[from-pstn]
; This is the context that would be listed in the config file
; for the circuit (i.e. chan_dahdi.conf)
exten = _X.,1,Dial(SIP/android-esteban)
[number-mapping]
; This context is not strictly required, but will make it easier
; to keep track of your DIDs in a single location in your dialplan.
; From here you can pass the call to another part of the dialplan
; where the actual dialplan work will take place.
exten = 59521xx,1,Dial(SIP/android-esteban)
exten = i,1,Verbose(2,Incoming call to invalid number)
Dahdi system.conf
# Autogenerated by /usr/sbin/dahdi_genconf on Tue Sep 6 14:40:03 2011
# If you edit this file and execute /usr/sbin/dahdi_genconf again,
# your manual changes will be LOST.
# Dahdi Configuration File
#
# This file is parsed by the Dahdi Configurator, dahdi_cfg
#
# Span 1: WCTDM/4 Wildcard TDM400P REV I Board 5 (MASTER)
fxsks=1
echocanceller=mg2,1
# channel 2, WCTDM/4/1, no module.
# channel 3, WCTDM/4/2, no module.
# channel 4, WCTDM/4/3, no module.
# Global data
loadzone= us
defaultzone = us
Thanks in advance !
--
Esteban L. Cacavelos de Amoriza
Cel: 0981 220 429
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[asterisk-users] Beginner Question: Remote access
Hello list, I am a beginner at asterisk. I want to access my asterisk box from my laptop, on a different network (mobile hotspot). The asterisk box doesn't have a static ip, how do I connect with it using ssh or other such programs? Thanks for your guidance guys. It is highly 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] Beginner Question: Remote access
Google for IP-tunneling. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of A Dunor Sent: Tuesday, September 06, 2011 7:17 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Beginner Question: Remote access Hello list, I am a beginner at asterisk. I want to access my asterisk box from my laptop, on a different network (mobile hotspot). The asterisk box doesn't have a static ip, how do I connect with it using ssh or other such programs? Thanks for your guidance guys. It is highly 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 -- _ -- 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] Beginner Question: 4 fxo TDM410 setup
Hello list. Just another beginner question. I am trying to setup a basic home phone system. I ordered a TDM410 card, which came with 4 fxo ports. I want the home phone system to be able to initiate and receive calls. Can it be done with this card with just one type(no fxs) of ports? If it can be done please help me with scenarios. Thanks for your guidance guys. It is highly 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] Beginner Question: Remote access
Thanks for the speedy pointer Danny. On 9/6/2011 8:27 PM, Danny Nicholas wrote: Google for IP-tunneling. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of A Dunor Sent: Tuesday, September 06, 2011 7:17 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Beginner Question: Remote access Hello list, I am a beginner at asterisk. I want to access my asterisk box from my laptop, on a different network (mobile hotspot). The asterisk box doesn't have a static ip, how do I connect with it using ssh or other such programs? Thanks for your guidance guys. It is highly 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 -- _ -- 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] ISDN2 PCIe Card for Asterisk
yes, sangoma card is good. Best regards, James.zhu Doing asterisk/PRI/ss7/dahdi, linux, asterisk cards, gateway(fxs/fxo/pri-SIP). website: www.voipviews.com Date: Tue, 6 Sep 2011 10:38:44 +0200 From: th9...@googlemail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] ISDN2 PCIe Card for Asterisk what do you mean exactly?! One what?! What do you plan to accomplish?! Do you mean a 1 Port ISDN BRI Board?! Difficult to find, and thus boards are really expensive, not under 400.- � inkluding DSP Processors. I advise you taking Gentoo Linux, getting asterisk on it and put a single Port HFC-S PCI (not PCIe) Board in your CPU. If you need something really professional, for Serverside, I advise you sangoma. Tamer Am 06.09.2011 09:08, schrieb Arjan Kroon | Mobillion: Hi, I'm looking for a PCIe card with 1 ISDN2 connection which works with Asterisk Could anybody give me an advise which card I can use? Regards, Arjan Kroon Mobillion. -- _ -- 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] Beginner Question: 4 fxo TDM410 setup
hi: yes, 4fxo is enough for that. four fox means that you have 4 PSTN line, do you really need 4 fxos? you have to use fxs or sip as extensions for pick up the call and make calls. Best regards, James.zhu Doing asterisk/PRI/ss7/dahdi, linux, asterisk cards, gateway(fxs/fxo/pri-SIP). website: www.voipviews.com Date: Tue, 6 Sep 2011 20:28:05 -0400 From: alsta...@gmail.com To: asterisk-users@lists.digium.com Subject: [asterisk-users] Beginner Question: 4 fxo TDM410 setup Hello list. Just another beginner question. I am trying to setup a basic home phone system. I ordered a TDM410 card, which came with 4 fxo ports. I want the home phone system to be able to initiate and receive calls. Can it be done with this card with just one type(no fxs) of ports? If it can be done please help me with scenarios. Thanks for your guidance guys. It is highly 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 -- _ -- 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] trying to build 1.8.6.0 on CentOS 6, problems with ptlib
On 06/09/11 05:14 PM, Kevin P. Fleming wrote: I was trying ./configure --disable-chan_ooh323 and that was not making a difference. It won't, for two reasons: Asterisk modules can't be selected/deselected via the configure script (menuselect is used for that), and chan_ooh323 doesn't use pwlib/openh323, chan_h323 does. However you could select/deselect modules using menuselect if you wanted to automate the process. It's documented over here: http://ofps.oreilly.com/titles/9780596517342/asterisk-Install.html#Installing_id293439 (Just thought I'd pass that along as I thought it was pretty neat when I learned about it :)) -- Leif Madsen http://www.oreilly.com/catalog/asterisk -- _ -- 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 subject)
Hi team,
I am trying to find solution to hangup b-party call after 1 min with out
disconnecting the call of a-party but following dial plan which is disconnect
both the calls.
Please suggest me the solution.
[TB]
exten = _X.,1,Wait(${INCOMING_WAIT})
exten =_X.,2,Verbose(TB)
exten =_X.,3,Answer()
exten = _X.,4,Set(mainLoop=0)
exten = _X.,5,Set(TIMEOUT(absolute)=10) ; set time in milliseconds
exten = _X.,6,Playback(/var/callagent/prompts/monitor/thanks)
exten = _X.,7,Dial(DAHDI/7/
09501032209,10,S(60))
exten = _X.,8,Noop(${DIALEDTIME})
exten =_X.,9,Goto(TB,_X.,1)
exten =_X.,n,Hangup()
Thanks
Vinod Dharashive
Sent from BlackBerry® on Airtel
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Re: [asterisk-users] Beginner Question: Remote access
There could be as easy solutions as using teamviewer or use tools like Hamachi used in combination with dyn-dns etc. IP-tunneling I guess needs static public IPs for the sake of completing the route. On Wed, Sep 7, 2011 at 5:30 AM, A Dunor alsta...@gmail.com wrote: Thanks for the speedy pointer Danny. On 9/6/2011 8:27 PM, Danny Nicholas wrote: Google for IP-tunneling. -Original Message- From: asterisk-users-bounces@lists.**digium.comasterisk-users-boun...@lists.digium.com [mailto:asterisk-users-**boun...@lists.digium.comasterisk-users-boun...@lists.digium.com] On Behalf Of A Dunor Sent: Tuesday, September 06, 2011 7:17 PM To: asterisk-users@lists.digium.**com asterisk-users@lists.digium.com Subject: [asterisk-users] Beginner Question: Remote access Hello list, I am a beginner at asterisk. I want to access my asterisk box from my laptop, on a different network (mobile hotspot). The asterisk box doesn't have a static ip, how do I connect with it using ssh or other such programs? Thanks for your guidance guys. It is highly 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-**usershttp://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-**usershttp://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-**usershttp://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] Beginner Question: Remote access
On Tue, Sep 06, 2011 at 08:17:01PM -0400, A Dunor wrote: Hello list, I am a beginner at asterisk. I want to access my asterisk box from my laptop, on a different network (mobile hotspot). The asterisk box doesn't have a static ip, how do I connect with it using ssh or other such programs? If it has an external IP address but not a static one, you can provide it a permaninet hostname using services such as dyndns.com . Alternatively (if it is behind NAT), if you have any external host under your control, you can create a tunnel (e.g.: openvpn) between that host and your Asterisk server. -- 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 -- 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] Queue agent login notification
We're using FreePBX and I'm wondering if it's possible to add to the login/logout macros a command to execute an AGI/Command to launch an external process for that. Thanks. On Fri, Aug 12, 2011 at 2:30 PM, Alex Vishnev alex9...@gmail.com wrote: you can monitor queue_log file for ADDMEMBER or REMOVEMEMBER events. from that point on, you can store them or take any other action. the other way is to use AMI an monitor for Agent login/logoff events On Aug 12, 2011, at 7:06 AM, Michael wrote: Hello, Is there a way to either store login/logout agent information in a database or at least send an email when an agent logs in or out of a queue? Thanks, Michael -- _ -- 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] Queue agent login notification
See this link: http://www.voip-info.org/wiki/view/Asterisk+queue_log+on+MySQL You'll find similar pages where you can setup to store queue logs/events(as Alex mentioned) in MySQL DB and further do your triggers or functions on them. On Wed, Sep 7, 2011 at 10:46 AM, Michael voip.quest...@gmail.com wrote: We're using FreePBX and I'm wondering if it's possible to add to the login/logout macros a command to execute an AGI/Command to launch an external process for that. Thanks. On Fri, Aug 12, 2011 at 2:30 PM, Alex Vishnev alex9...@gmail.com wrote: you can monitor queue_log file for ADDMEMBER or REMOVEMEMBER events. from that point on, you can store them or take any other action. the other way is to use AMI an monitor for Agent login/logoff events On Aug 12, 2011, at 7:06 AM, Michael wrote: Hello, Is there a way to either store login/logout agent information in a database or at least send an email when an agent logs in or out of a queue? Thanks, Michael -- _ -- 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
