[asterisk-users] Wrong call information on B leg
Greetings.
I have next feature in features.conf :
send =
*9,peer/both,AGI,/etc/asterisk/agi/map_mail.pl
What it does is parsing
CALLERID and DNID from AGI input, performing some actions in MySQL with
these values, and then running application for peer (for example,
PlayBack)
Sounds simple, and it really is. When my user is receiving a
call (we are the B leg) and presses *9, everything works perfectly.
However, there is a problem if we are the A leg. For some reason,
variables parsed from AGI input are not real for current call.
For
example, my SIP user 405 calls PSTN number, let's s say 044201, via
PRI.
AGI debug output in console shows next:
AGI Tx agi_request:
/etc/asterisk/agi/map_mail.pl
AGI Tx agi_channel: Zap/63-1
AGI Tx
agi_language: en
AGI Tx agi_type: Zap
AGI Tx agi_uniqueid:
1322049810.4307
AGI Tx agi_callerid: 044201
AGI Tx
agi_calleridname: unknown
AGI Tx agi_callingpres: 3
AGI Tx
agi_callingani2: 0
AGI Tx agi_callington: 0
AGI Tx
agi_callingtns: 0
AGI Tx agi_dnid: 481
AGI Tx agi_rdnis:
unknown
AGI Tx agi_context: from_pstn
AGI Tx agi_extension:
AGI Tx agi_priority: 1
AGI Tx agi_enhanced: 0.0
AGI Tx
agi_accountcode:
As we can see here, the DNID and CALLERID are swapped
(idk, maybe it is intended behavior). However, that is not a problem.
The problem is - it shows 481, but we are calling from 405!
Okay, I
said, I will pass the ${DNID} and ${CALLERID} to my script as
parameters. Then I run SHOW CHANNEL on my B leg (i.e. Zap/63-1, opposed
to SIP/405-somecallid) to see what these variables are, and I see next:
Caller ID: 044201
Caller ID Name: (N/A)
DNID Digits: 408
(it
is another call, so DNID differs, but it is still not 405)
As far as I
understand from manuals, by setting peer in my features.conf I make
script being launched for the other side from user, who pressed *9 - on
B leg, if the call is outgoing.
And as I can see from what I get here,
script receives information for its leg - and gets the wrong information
in situation described above.
The question is - is this really the way
Asterisk works with channel variables, or am I doing something wrong?
And if the problem is me - how can I receive 100% correct information
for current call, not depending on if it is launched on caller or callee
side?
My Asterisk version is Asterisk SVN-branch-1.4-r290100 and I
really hope that I will not have to update it because it will be a long
and hard quest due to old and broken PRI card drivers 8(
Hope for some
wisdom and help.
--
With Best Regards
Mikhail Lischuk
ITX Ukraine
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Re: [asterisk-users] Wrong call information on B leg
I am out of the office until 12/16 but I will still be checking my messages. For immediate assistance, please call Telecommunication Services at 799-6543. Thank you. Chet Stevens Telecommunication Services Clark County School District -- _ -- 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] how to get the Record_ID
Hello List coud you please show me how to get the RECORD_ID for all outbond calls, i use asterisk 1.4 with database mysql thanks and 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
[asterisk-users] Digium TE205P leds flash red on startup
Hi, I have a new Digium TE205P 2-span E1 card I just installed on a server. As soon as I boot the machine, the card's leds flash red (ports 1 and 2) - even when in the BIOS. That's not good, right? I don't have another machine to test at the moment but would like to know what to expect. I have several single-span E1 cards and when the machine boots, their leds are off until the kernel module is loaded. What could be the problem with my TE205P? Could it be damaged (brand new) or is it more likely to be a PCI-BIOS issue? Thanks, Vieri -- _ -- 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] Digium TE205P leds flash red on startup
On Thu, Dec 15, 2011 at 11:05 AM, Vieri rentor...@yahoo.com wrote: Hi, I have a new Digium TE205P 2-span E1 card I just installed on a server. As soon as I boot the machine, the card's leds flash red (ports 1 and 2) - even when in the BIOS. That's not good, right? I don't have another machine to test at the moment but would like to know what to expect. I have several single-span E1 cards and when the machine boots, their leds are off until the kernel module is loaded. What could be the problem with my TE205P? Could it be damaged (brand new) or is it more likely to be a PCI-BIOS issue? Thanks, Vieri That is normal expected behavior. The card is in a red alarm state that just means there is no link. Its fine. -- ~ Andrew lathama Latham lath...@gmail.com http://lathama.net ~ -- _ -- 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] app_swift tts module - new home.
Hi Folks, After receiving a surprising amount of emails from Asterisk community members, I thought I'd fire something off to the users list to clear any confusion regarding the Asterisk Forge (forge.asterisk.org) website and the future of the app_swift text-to-speech module. With regards to the Asterisk Forge website redirecting to GitHub, this has been a long time coming. Emails were sent out to the various lists warning folks that the hosted GForge site was going away - so no one should be too surprised - 'nuf said there. As far as the app_swift project is concerned, with the exception of moving things around as far as location, it is business as usual. The app_swift code for *all* the different versions of Asterisk is now being hosted on GitHub at https://github.com/dmsessions/app_swift . This is a good thing and will make life easier. btw, I love git. If you don't yet, you will . . someday soon . . Individual tar files for each of the different versions of app_swift, which is what 99% of people are going to want, are all available for download on my website at http://www.darrensessions.com by clicking the 'Downloads' button at the very top of the page. That is all my friends. Seasons Greetings! - Darren -- _ -- 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] Partner Keys on Innovaphone
Hello, when using BLF with Asterisk 1.6, I notice that the Caller-ID information is not displayed on the monitoring key of my Innovaphone IP200A. If the IP-phone of my colleague rings, I should see on my partner key the number of the caller. This is information that is being send in the xml-body of the NOTIFY-message. I do not see this information in the xml-body of a NOTICE-message from the Asterisk PBX. Is it not supported ? Is it supported in 1.8 ? 1.10 ? Kind regards, Jonas. -- _ -- 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] Partner Keys on Innovaphone
AIR this is 10.0 functionality; to make it work in the 1.X tree you have to blind transfer From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens Sent: Thursday, December 15, 2011 9:42 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Partner Keys on Innovaphone Hello, when using BLF with Asterisk 1.6, I notice that the Caller-ID information is not displayed on the monitoring key of my Innovaphone IP200A. If the IP-phone of my colleague rings, I should see on my partner key the number of the caller. This is information that is being send in the xml-body of the NOTIFY-message. I do not see this information in the xml-body of a NOTICE-message from the Asterisk PBX. Is it not supported ? Is it supported in 1.8 ? 1.10 ? Kind regards, Jonas. -- _ -- 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] Partner Keys on Innovaphone
But blind transfer has nothing to do with BLF, no ?! It should work with a normal incoming call, not just a transfer. Jonas. On 12/15/2011 04:42 PM, Danny Nicholas wrote: AIR this is 10.0 functionality; to make it work in the 1.X tree you have to blind transfer *From:*asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonas Kellens *Sent:* Thursday, December 15, 2011 9:42 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Partner Keys on Innovaphone Hello, when using BLF with Asterisk 1.6, I notice that the Caller-ID information is not displayed on the monitoring key of my Innovaphone IP200A. If the IP-phone of my colleague rings, I should see on my partner key the number of the caller. This is information that is being send in the xml-body of the NOTIFY-message. I do not see this information in the xml-body of a NOTICE-message from the Asterisk PBX. Is it not supported ? Is it supported in 1.8 ? 1.10 ? Kind regards, Jonas. -- _ -- 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] Partner Keys on Innovaphone
BLF as I understand it is controlled by hints; perhaps I am crossing wires here. But your usage as stated expects the caller ID of a call to go into the hint information Colleague is leg B Caller is leg A You are looking for hint to be populated when leg B is established. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens Sent: Thursday, December 15, 2011 9:47 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Partner Keys on Innovaphone But blind transfer has nothing to do with BLF, no ?! It should work with a normal incoming call, not just a transfer. Jonas. On 12/15/2011 04:42 PM, Danny Nicholas wrote: AIR this is 10.0 functionality; to make it work in the 1.X tree you have to blind transfer From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens Sent: Thursday, December 15, 2011 9:42 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Partner Keys on Innovaphone Hello, when using BLF with Asterisk 1.6, I notice that the Caller-ID information is not displayed on the monitoring key of my Innovaphone IP200A. If the IP-phone of my colleague rings, I should see on my partner key the number of the caller. This is information that is being send in the xml-body of the NOTIFY-message. I do not see this information in the xml-body of a NOTICE-message from the Asterisk PBX. Is it not supported ? Is it supported in 1.8 ? 1.10 ? Kind regards, Jonas. -- _ -- 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] server unresponsive
Hi,
I had some problems with sip peers losing connection suddenly without real
network issues. In my case, it was useful to refresh the table used for
real time configuration, we made an script for Postgres like this
PGUSER=user PGPASSWORD=password vacuumdb --full --table 'sip_buddies'
asterisk_db
Greetings,
Elder D. Arohuanca
On Fri, Nov 20, 2009 at 6:33 PM, Edwin Lam edwin@officegeneral.comwrote:
Paul Scott wrote:
How is your network structured?
we have a central location where the * server is located. and 4 remote
locations connected via point to point lines.
Can you show me a sample entry from your sip.conf?
we use realtime sip w/ mysql tables. a typical entry would looks
like this:
[1234]
context=default10
type=friend
secret=xyz
qualify=yes
host=dynamic
canreinvite=no
dtmfmode=rfc2833
disallow=all
allow=g729
allow=alaw
allow=ulaw
I was having this problem. But as far as I could tell there wasn't
one. From a network stand point all phones were reachable asterisk
was just reporting that it was Unreachable and it wasn't sending the
calls. I switched to qualify=no and wrote a small agi to catch $
{HANGUPCAUSE} and log it to a file. If it records a bunch of
chanunavail messages you still have a problem.
If you don't want to turn qualify off you could play with the qualify
times. I did a bunch of this before I just gave up.
I'm sure there is a better or proper way of handling this. I'm
interested to hear it.
Paul
On Nov 20, 2009, at 3:41 PM, Edwin Lam wrote:
hi folks.
we've experienced some weird problems lately. we have about 600
SIP phone on a single system running *1.4.26.2 for about a month.
recently there was massive UNREACHABLE messages like this one
showed up:
chan_sip.c: Peer '2699' is now UNREACHABLE! Last qualify: 1252
then they all became reachable again in a few seconds. sometimes
it last for couple minutes. but sometimes it last for hours, when
that happens. the system will get very slow and eventually error
like this will start showing:
channel.c: Exceptionally long voice queue length queuing to IAX2/
hostpbx2-12619
after a while the whole system will become unresponsive
until i kill the asterisk process.
i've checked our network switches/routers and connections.
they all work fine without any packet lost.
any suggestions?
--
Edwin Lam edwin@officegeneral.com
Systems Engineer, Office General, Inc.
Ph: +1 415 439 4988 Fax: +1 415 283 3370
http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20
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Re: [asterisk-users] Digium TE205P leds flash red on startup
- Original Message - From: Vieri rentor...@yahoo.com As soon as I boot the machine, the card's leds flash red (ports 1 and 2) - even when in the BIOS. That's not good, right? This is normal, no need to be concerned. -- _ -- 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] Partner Keys on Innovaphone
If Asterisk supports rfc 4235, then it should be able to send this caller-information inside the NOTIFY-message, but it does not. So my question is : does Asterisk 1.6 not support this ? Or am I lacking something in my configuration ? Jonas. On 12/15/2011 04:51 PM, Danny Nicholas wrote: BLF as I understand it is controlled by hints; perhaps I am crossing wires here. But your usage as stated expects the caller ID of a call to go into the hint information Colleague is leg B Caller is leg A You are looking for hint to be populated when leg B is established. *From:*asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonas Kellens *Sent:* Thursday, December 15, 2011 9:47 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Partner Keys on Innovaphone But blind transfer has nothing to do with BLF, no ?! It should work with a normal incoming call, not just a transfer. Jonas. On 12/15/2011 04:42 PM, Danny Nicholas wrote: AIR this is 10.0 functionality; to make it work in the 1.X tree you have to blind transfer *From:*asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonas Kellens *Sent:* Thursday, December 15, 2011 9:42 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Partner Keys on Innovaphone Hello, when using BLF with Asterisk 1.6, I notice that the Caller-ID information is not displayed on the monitoring key of my Innovaphone IP200A. If the IP-phone of my colleague rings, I should see on my partner key the number of the caller. This is information that is being send in the xml-body of the NOTIFY-message. I do not see this information in the xml-body of a NOTICE-message from the Asterisk PBX. Is it not supported ? Is it supported in 1.8 ? 1.10 ? Kind regards, Jonas. -- _ -- Bandwidth and Colocation Provided byhttp://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
[asterisk-users] Play audio file for both Caller and Callee in a call
Dear all, Anyone of you knows how to play an audio file at the beginning of a call for both Caller and Callee? A(x) of Dial application only plays audio for callee. I don't want to use MeetMe because I want to use Monitor and MixMonitor. Thank you! Este mensaje se dirige exclusivamente a su destinatario. Puede consultar nuestra pol?tica de env?o y recepci?n de correo electr?nico en el enlace situado m?s abajo. This message is intended exclusively for its addressee. We only send and receive email on the basis of the terms set out at. http://www.tid.es/ES/PAGINAS/disclaimer.aspx -- _ -- 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] Play audio file for both Caller and Callee in a call
Playback? What flavor of Asterisk are you using? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of ISABEL ORDAS ARNAL Sent: Thursday, December 15, 2011 10:29 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Play audio file for both Caller and Callee in a call Dear all, Anyone of you knows how to play an audio file at the beginning of a call for both Caller and Callee? A(x) of Dial application only plays audio for callee. I dont want to use MeetMe because I want to use Monitor and MixMonitor. Thank you! _ Este mensaje se dirige exclusivamente a su destinatario. Puede consultar nuestra política de envío y recepción de correo electrónico en el enlace situado más abajo. This message is intended exclusively for its addressee. We only send and receive email on the basis of the terms set out at. http://www.tid.es/ES/PAGINAS/disclaimer.aspx -- _ -- 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] Digium TE205P leds flash red on startup
On Thursday 15 December 2011, Vieri wrote: Hi, I have a new Digium TE205P 2-span E1 card I just installed on a server. As soon as I boot the machine, the card's leds flash red (ports 1 and 2) - even when in the BIOS. That's not good, right? No, it's normal behaviour until the card's firmware has been loaded. Which can't happen until the kernel is booted; and probably will not happen until the Zaptel or DAHDI startup script runs. -- AJS Answers come *after* questions. -- _ -- 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] Play audio file for both Caller and Callee in a call
Dear Danny: How can you use Playback in the middle of 2 channels engaged in a conversation?ThanksC. Savinovich Original Message Subject: Re: [asterisk-users] Play audio file for both Caller and Callee in a call From: "Danny Nicholas" da...@debsinc.com Date: Thu, December 15, 2011 9:31 am To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" asterisk-users@lists.digium.com Playback? What flavor of Asterisk are you using?From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of ISABEL ORDAS ARNALSent: Thursday, December 15, 2011 10:29 AMTo: asterisk-users@lists.digium.comSubject: [asterisk-users] Play audio file for both Caller and Callee in a callDear all, Anyone of you knows how to play an audio file at the beginning of a call for both Caller and Callee?A(x) of Dial application only plays audio for callee. I don’t want to use MeetMe because I want to use Monitor and MixMonitor. Thank you!Este mensaje se dirige exclusivamente a su destinatario. Puede consultar nuestra política de envío y recepción de correo electrónico en el enlace situado más abajo.This message is intended exclusively for its addressee. We only send and receive email on the basis of the terms set out at.http://www.tid.es/ES/PAGINAS/disclaimer.aspx-- _ -- 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] Play audio file for both Caller and Callee in a call
You can’t per se, but you can call an AGI using stream? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of c.savinov...@itntelecom.com Sent: Thursday, December 15, 2011 11:22 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Play audio file for both Caller and Callee in a call Dear Danny: How can you use Playback in the middle of 2 channels engaged in a conversation? Thanks C. Savinovich Original Message Subject: Re: [asterisk-users] Play audio file for both Caller and Callee in a call From: Danny Nicholas mailto:da...@debsinc.com da...@debsinc.com Date: Thu, December 15, 2011 9:31 am To: 'Asterisk Users Mailing List - Non-Commercial Discussion' mailto:asterisk-users@lists.digium.com asterisk-users@lists.digium.com Playback? What flavor of Asterisk are you using? From: mailto:asterisk-users-boun...@lists.digium.com asterisk-users-boun...@lists.digium.com [ mailto:asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of ISABEL ORDAS ARNAL Sent: Thursday, December 15, 2011 10:29 AM To: mailto:asterisk-users@lists.digium.com asterisk-users@lists.digium.com Subject: [asterisk-users] Play audio file for both Caller and Callee in a call Dear all, Anyone of you knows how to play an audio file at the beginning of a call for both Caller and Callee? A(x) of Dial application only plays audio for callee. I don’t want to use MeetMe because I want to use Monitor and MixMonitor. Thank you! _ Este mensaje se dirige exclusivamente a su destinatario. Puede consultar nuestra política de envío y recepción de correo electrónico en el enlace situado más abajo. This message is intended exclusively for its addressee. We only send and receive email on the basis of the terms set out at. http://www.tid.es/ES/PAGINAS/disclaimer.aspx http://www.tid.es/ES/PAGINAS/disclaimer.aspx _ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users 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] Play audio file for both Caller and Callee in a call
You also use AMI to inject audio into the conversation using the ChanSpy application. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Dec 15, 2011, at 9:23 AM, Danny Nicholas wrote: You can’t per se, but you can call an AGI using stream? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of c.savinov...@itntelecom.com Sent: Thursday, December 15, 2011 11:22 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Play audio file for both Caller and Callee in a call Dear Danny: How can you use Playback in the middle of 2 channels engaged in a conversation? Thanks C. Savinovich Original Message Subject: Re: [asterisk-users] Play audio file for both Caller and Callee in a call From: Danny Nicholas da...@debsinc.com Date: Thu, December 15, 2011 9:31 am To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Playback? What flavor of Asterisk are you using? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of ISABEL ORDAS ARNAL Sent: Thursday, December 15, 2011 10:29 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Play audio file for both Caller and Callee in a call Dear all, Anyone of you knows how to play an audio file at the beginning of a call for both Caller and Callee? A(x) of Dial application only plays audio for callee. I don’t want to use MeetMe because I want to use Monitor and MixMonitor. Thank you! Este mensaje se dirige exclusivamente a su destinatario. Puede consultar nuestra política de envío y recepción de correo electrónico en el enlace situado más abajo. This message is intended exclusively for its addressee. We only send and receive email on the basis of the terms set out at. http://www.tid.es/ES/PAGINAS/disclaimer.aspx -- _ -- 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
[asterisk-users] Asterisk log format
Hi mates! Please, I need to understand how to search for an specific log by date/time on asterisk logs, but can't understand how this works, can you guys please give me an example about how those logs works? Best regards, Asterisk Guy -- _ -- 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] Best PBX for Call Centers?
Hello List, I have customer with a 40 Agents call center. and is looking to install a PBX switch that can serve those agents. As per my experience i suggested Asterisk as i have tested it with Call Centers, however he has been advised not to use it although his provider is using Asterisk to send him calls. He has been advised to use Sippy which they claim is more stable than Asterisk. i'm not an expert with Sippy so i'm looking for a piece of an advise here.. if i'm doing an Asterisk Vs Sippy comparison. can anyone help? Regards Tarek Sawah Information Technology Adviser Integrated Digital Systems CCNP, MCSE, RHCE, TELECOM USA: +1 386 492 9993 -- _ -- 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] Best PBX for Call Centers?
On Thu, Dec 15, 2011 at 11:33 AM, Tarek Sawah tareksa...@hotmail.comwrote: Hello List, I have customer with a 40 Agents call center. and is looking to install a PBX switch that can serve those agents. As per my experience i suggested Asterisk as i have tested it with Call Centers, however he has been advised not to use it although his provider is using Asterisk to send him calls. He has been advised to use Sippy which they claim is more stable than Asterisk. More stable? We have Asterisk servers that have run for many years without being unstable. There's a pair of them in a colo facility that are 7 years old and haven't been touched in at least two years. Just how many more years do you need to be more stable? Asterisk isn't perfect, but done right it's quite stable. IMO, the people giving you advice just don't know how to do it right. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- 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] Asterisk log format
grep or sgrep Sent from my iPhone On 15 dec 2011, at 18:46, Asterisk Guy arpexpe...@gmail.com wrote: Hi mates! Please, I need to understand how to search for an specific log by date/time on asterisk logs, but can't understand how this works, can you guys please give me an example about how those logs works? Best regards, Asterisk Guy -- _ -- 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] Best PBX for Call Centers?
On 12/15/2011 01:33 PM, Tarek Sawah wrote: Hello List, I have customer with a 40 Agents call center. and is looking to install a PBX switch that can serve those agents. As per my experience i suggested Asterisk as i have tested it with Call Centers, however he has been advised not to use it although his provider is using Asterisk to send him calls. He has been advised to use Sippy which they claim is more stable than Asterisk. i'm not an expert with Sippy so i'm looking for a piece of an advise here.. if i'm doing an Asterisk Vs Sippy comparison. can anyone help? Regards I think the answer you're going to get on an Asterisk mailing list is Asterisk is the best. I'd find out why your customer is being advised against Asterisk. Bet you'll find political reasons rather than technical. And if they are technical reasons, make sure they're applicable to a recent version of Asterisk rather than hearing It doesn't do Feature X and then finding out that Feature X was added in 1.4 (and the Asterisk world is on the equivalent of 1.10) and the person just has experience with 1.2. -- _ -- 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] Best PBX for Call Centers?
On Thu, Dec 15, 2011 at 3:39 PM, Carlos Alvarez car...@televolve.com wrote: On Thu, Dec 15, 2011 at 11:33 AM, Tarek Sawah tareksa...@hotmail.com wrote: Hello List, I have customer with a 40 Agents call center. and is looking to install a PBX switch that can serve those agents. As per my experience i suggested Asterisk as i have tested it with Call Centers, however he has been advised not to use it although his provider is using Asterisk to send him calls. He has been advised to use Sippy which they claim is more stable than Asterisk. More stable? We have Asterisk servers that have run for many years without being unstable. There's a pair of them in a colo facility that are 7 years old and haven't been touched in at least two years. Just how many more years do you need to be more stable? Asterisk isn't perfect, but done right it's quite stable. IMO, the people giving you advice just don't know how to do it right. -- Carlos Alvarez TelEvolve 602-889-3003 I fully support the statement by Carlos. Planing, engineering and other factors can make almost any software stable. Experience is key. -- ~ Andrew lathama Latham lath...@gmail.com http://lathama.net ~ -- _ -- 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] Best PBX for Call Centers?
On 12/15/2011 01:43 PM, James Sharp wrote: On 12/15/2011 01:33 PM, Tarek Sawah wrote: Hello List, I have customer with a 40 Agents call center. and is looking to install a PBX switch that can serve those agents. As per my experience i suggested Asterisk as i have tested it with Call Centers, however he has been advised not to use it although his provider is using Asterisk to send him calls. He has been advised to use Sippy which they claim is more stable than Asterisk. i'm not an expert with Sippy so i'm looking for a piece of an advise here.. if i'm doing an Asterisk Vs Sippy comparison. can anyone help? Regards I think the answer you're going to get on an Asterisk mailing list is Asterisk is the best. I'd find out why your customer is being advised against Asterisk. Bet you'll find political reasons rather than technical. And if they are technical reasons, make sure they're applicable to a recent version of Asterisk rather than hearing It doesn't do Feature X and then finding out that Feature X was added in 1.4 (and the Asterisk world is on the equivalent of 1.10) and the person just has experience with 1.2. I should have probably read the whole message. Especially the part about the more stable claim. -- _ -- 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] Digium TE205P leds flash red on startup
--- On Thu, 12/15/11, A J Stiles asterisk_l...@earthshod.co.uk wrote: I have a new Digium TE205P 2-span E1 card I just installed on a server. As soon as I boot the machine, the card's leds flash red (ports 1 and 2) - even when in the BIOS. That's not good, right? No, it's normal behaviour until the card's firmware has been loaded. Which can't happen until the kernel is booted; and probably will not happen until the Zaptel or DAHDI startup script runs. Well, strange enough, the server used to have a single-span PRI card, booted with kernel 2.6.23 and autoloaded the appropriate zaptel 1.4.12.1 module (wcte12xp). Now I replaced the single-span card with the dual-span TE205 and rebooted. The kernel does not autoload the new zaptel module which should be wct4xxp. So I try to load it manually (modprobe -a wct4xxp) and lsmod lists it but there's nothing in /proc/zaptel/. I suppose the 1205 identifier is correct for the TE205 card, as seen after issuing lspci: 05:01.0 Communication controller: Digium, Inc. Unknown device 1205 (rev 02) Subsystem: Unknown device 0005: Flags: bus master, medium devsel, latency 64, IRQ 5 Memory at feaefc00 (32-bit, non-prefetchable) [size=128] I left my zaptel.conf and zapata.conf files untouched as, theoretically, they should work just fine, at least for the first PRI port on the card (everything else is identical). So zaptel.conf has something like this: span=1,1,0,ccs,hdb3,crc4 bchan=1-15 dchan=16 bchan=17-31 and zapata.conf: switchtype = euroisdn signalling = pri_cpe group = 1 channel = 1-15 channel = 17-31 However, if I run ztcfg I get this message: ZT_SPANCONFIG failed on span 1: No such device or address (6) The fact that there's nothing in /proc/zaptel/ makes me think that the zaptel kernel module isn't working. Is the 1205 card compatible with zaptel 1.4.12.1? (I can't migrate to DAHDI on this system - at least not yet) Thanks, Vieri -- _ -- 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] Best PBX for Call Centers?
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andrew Latham Sent: Thursday, December 15, 2011 1:46 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Best PBX for Call Centers? On Thu, Dec 15, 2011 at 3:39 PM, Carlos Alvarez car...@televolve.com wrote: On Thu, Dec 15, 2011 at 11:33 AM, Tarek Sawah tareksa...@hotmail.com wrote: Hello List, I have customer with a 40 Agents call center. and is looking to install a PBX switch that can serve those agents. As per my experience i suggested Asterisk as i have tested it with Call Centers, however he has been advised not to use it although his provider is using Asterisk to send him calls. He has been advised to use Sippy which they claim is more stable than Asterisk. More stable? We have Asterisk servers that have run for many years without being unstable. There's a pair of them in a colo facility that are 7 years old and haven't been touched in at least two years. Just how many more years do you need to be more stable? Asterisk isn't perfect, but done right it's quite stable. IMO, the people giving you advice just don't know how to do it right. I fully support the statement by Carlos. Planing, engineering and other factors can make almost any software stable. Experience is key. Reply: I my experience, once you have a version of Asterisk which is stable for you DO NOT UPDATE unless you have NO other choice. We used to apply updates but got burned far too many times, usually by bugs which would not show up during normal testing. Now we only update for serious security issue, and even then we try to handle the issue some other way. -- _ -- 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] Best PBX for Call Centers?
On Thu, Dec 15, 2011 at 11:52 AM, Eric Wieling ewiel...@nyigc.com wrote: I my experience, once you have a version of Asterisk which is stable for you DO NOT UPDATE unless you have NO other choice. We used to apply updates but got burned far too many times, usually by bugs which would not show up during normal testing. Now we only update for serious security issue, and even then we try to handle the issue some other way. Same here. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- 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] Call pickup on Asterisk 1.8 and Polycom IP550s?
This is one of those Is anyone else doing this?/Is anyone else seeing this? posts. We have an Asterisk 1.8.4 system, with Polycom IP550 phones running firmware 3.2.3. If someone on the 'buddy list' - the list of other extensions to watch - is called, the phone gets a NOTIFY event and displays a screen with the call information and a pickup softkey. However, if someone on that list is already on the phone and they get a second incoming call, the NOTIFY event comes in but the phone never displays the changed screen with the pickup button. It'll flash the light next to that extension, but that's it. Is anyone using a similar setup and seeing this? It's somewhat rare, but I have an office location where everyone there likes to pick up other people's calls, and they haven't been using a call queue like they oughta. Justin Sherrill - American Rock Salt P: 585-991-6825 F: 585-991-6925 C: 585-298-6826 -- _ -- 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] Call pickup on Asterisk 1.8 and Polycom IP550s?
AFAIK, Asterisk only picks up the first instance of a line, so if you have 2 calls on exten 100, only the first one is recognized. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Justin Sherrill Sent: Thursday, December 15, 2011 2:11 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Call pickup on Asterisk 1.8 and Polycom IP550s? This is one of those Is anyone else doing this?/Is anyone else seeing this? posts. We have an Asterisk 1.8.4 system, with Polycom IP550 phones running firmware 3.2.3. If someone on the 'buddy list' - the list of other extensions to watch - is called, the phone gets a NOTIFY event and displays a screen with the call information and a pickup softkey. However, if someone on that list is already on the phone and they get a second incoming call, the NOTIFY event comes in but the phone never displays the changed screen with the pickup button. It'll flash the light next to that extension, but that's it. Is anyone using a similar setup and seeing this? It's somewhat rare, but I have an office location where everyone there likes to pick up other people's calls, and they haven't been using a call queue like they oughta. Justin Sherrill - American Rock Salt P: 585-991-6825 F: 585-991-6925 C: 585-298-6826 -- _ -- 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] Digium TE205P leds flash red on startup
On 12/15/11 12:47 PM, Vieri wrote: ZT_SPANCONFIG failed on span 1: No such device or address (6) The fact that there's nothing in /proc/zaptel/ makes me think that the zaptel kernel module isn't working. Is the 1205 card compatible with zaptel 1.4.12.1? (I can't migrate to DAHDI on this system - at least not yet) Thanks, Vieri -- That's the reason why it's not working. Unfortunately, the newer versions of those cards require DAHDI in order to operate. Matthew Fredrickson Software/Hardware Engineer Digium, Inc. -- _ -- 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] Struggling with Extensions in Realtime
Hello Everyone, Can someone please let me know what the correct way to deal with extensions for a particular user using asterisk reatime. For a user 1001, we would like to support: Local Calls: 123-456-7890 LD Calls: 1-123-456-7890 INT Calls: 011-64-03-123-456-7890 PBX EXT:1002 Do I need to insert multiple records for use 1001, each pointing to the different extensions in the extensions table (i.e., local-context, ld-context, int-context, and pbx-context)? Thanks in Advance, Nick. -- _ -- 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 1.8.8.0 Now Available
The Asterisk Development Team is pleased to announce the release of Asterisk 1.8.8.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.8.8.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this release: * Updated SIP 484 handling; added Incomplete control frame When a SIP phone uses the dial application and receives a 484 Address Incomplete response, if overlapped dialing is enabled for SIP, then the 484 Address Incomplete is forwarded back to the SIP phone and the HANGUPCAUSE channel variable is set to 28. Previously, the Incomplete application dialplan logic was automatically triggered; now, explicit dialplan usage of the application is required. (Closes ASTERISK-17288. Reported by: Mikael Carlsson Tested by: Matthew Jordan Review: https://reviewboard.asterisk.org/r/1416/) * Prevent IAX2 from getting IPv6 addresses via DNS IAX2 does not support IPv6 and getting such addresses from DNS can cause error messages on the remote end involving bad IPv4 address casts in the presence of IPv6/IPv4 tunnels. (Closes issue ASTERISK-18090. Patched by Kinsey Moore) * Fix bad RTP media bridges in directmedia calls on peers separated by multiple Asterisk nodes. (Closes issue ASTERISK-18340. Reported by: Thomas Arimont. Closes issue ASTERISK-17725. Reported by: kwk. Tested by: twilson, jrose) * Fix crashes in ast_rtcp_write() (Closes issue ASTERISK-18570) Related issues that look like they are the same problem: (Issue ASTERISK-17560, ASTERISK-15406, ASTERISK-15257, ASTERISK-13334, ASTERISK-9977, ASTERISK-9716) Review: https://reviewboard.asterisk.org/r/1444/ Patched by: Russell Bryant * Fix for incorrect voicemail duration in external notifications. This patch fixes an issue where the voicemail duration was being reported with a duration significantly less than the actual sound file duration. (Closes ASTERISK-16981. Reported by: Mary Ciuciu, Byron Clark, Brad House, Karsten Wemheuer, KevinH Tested by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1443) * Prevent segfault if call arrives before Asterisk is fully booted. (Patched by alecdavis. https://reviewboard.asterisk.org/r/1407/) * Fix remote Crash Vulnerability in SIP channel driver (AST-2011-012) http://downloads.asterisk.org/pub/security/AST-2011-012.pdf * Fix locking order in app_queue.c which caused deadlocks (Closes issue ASTERISK-18101. Reported by Paul Rolfe, patched by Gregory Nietsky) (Closes issue ASTERISK-18487. Reported by Jason Legault, patched by Gregory Nietsky) * Fix regression in configure script for libpri capability checks (Closes issue ASTERISK-18687. Reported by norbert, patched by Richard Mudgett) * Prevent BLF subscriptions from causing deadlocks. (Closes issue ASTERISK-18663) Review: https://reviewboard.asterisk.org/r/1563/ * Fix deadlock if peer is destroyed while sending MWI notice. (Closes issue ASTERISK-18747) Reported by: Gregory Hinton Nietsky * Fix issue with setting defaultenabled on categories that are already enabled by default. (Closes issue ASTERISK-18738) Reported by: Paul Belanger * Don't crash on INFO automon request with no channel AST-2011-014. When automon was enabled in features.conf, it was possible to crash Asterisk by sending an INFO request if no channel had been created yet. * Fixed crash from orphaned MWI subscriptions in chan_sip This patch resolves the issue where MWI subscriptions are orphaned by subsequent SIP SUBSCRIBE messages. * Default to nat=yes; warn when nat in general and peer differ AST-2011-013. It is possible to enumerate SIP usernames when the general and user/peer nat settings differ in whether to respond to the port a request is sent from or the port listed for responses in the Via header. In 1.4 and For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.8.0 Thank you for your continued support of 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] Asterisk 10.0.0 Is Released!
The Asterisk Development Team is proud to announce the release of Asterisk 10.0.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ Asterisk 10 is the next major release series of Asterisk. It will be a Standard support release, similar to Asterisk 1.6.2. For more information about support time lines for Asterisk releases, see the Asterisk versions page: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions With the release of the Asterisk 10 branch, the preceding '1.' has been removed from the version number per the blog post available at http://blogs.digium.com/2011/07/21/the-evolution-of-asterisk-or-how-we-arrived-at-asterisk-10/ The release of Asterisk 10 would not have been possible without the support and contributions of the community. You can find an overview of the work involved with the 10.0.0 release in the summary: http://svn.asterisk.org/svn/asterisk/tags/10.0.0/asterisk-10.0.0-summary.txt A short list of available features includes: * T.38 gateway functionality has been added to res_fax. * Protocol independent out-of-call messaging support. Text messages not associated with an active call can now be routed through the Asterisk dialplan. SIP and XMPP are supported so far. * New highly optimized and customizable ConfBridge application capable of mixing audio at sample rates ranging from 8kHz-192kHz * Addition of video_mode option in confbridge.conf to provide basic video conferencing in the ConfBridge() dialplan application. * Support for defining hints has been added to pbx_lua. * Replacement of Berkeley DB with SQLite for the Asterisk Database (AstDB). * Much, much more! A full list of new features can be found in the CHANGES file. http://svn.asterisk.org/svn/asterisk/branches/10/CHANGES Also, when upgrading a system between major versions, it is imperative that you read and understand the contents of the UPGRADE.txt file, which is located at: http://svn.asterisk.org/svn/asterisk/branches/10/UPGRADE.txt Thank you for your continued support of 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] Which device auto-registered an extension?
Hi all,
In sip.conf:
[general]
regcontext = autoreg
[devabc]
regexten = 543
creates exten= 543,1,Noop(devabc) in context autoreg when devabc
registers. But I can't use exten= _5XX,2,Dial(SIP/${EXTEN}) in the
dialplan, because there's no device SIP/543. Now I know I can add a line
like exten= 543,2,Dial(SIP/devabc) for each and every device that uses
regexten, but it would be a lot cleaner to be able to use something like
Dial(SIP/${WHAT_GOES_HERE?}) instead.
So is there a way for the dialplan to determine which device caused SIP to
auto-register an extension?
--
Barry
--
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[asterisk-users] Asterisk console suddenly extremely verbose...
Hi there. I started the console today to reload the extensions.conf file ; only to be greeted with extremely verbose console. Seems related to the zaptel card: Example: Supervisory frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] N(R): 020 P/F: 1 0 bytes of data voip*CLI [ 00 01 01 2f ] voip*CLI Supervisory frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] N(R): 023 P/F: 1 0 bytes of data Handling message for SAPI/TEI=0/0 -- ACKing all packets from 22 to (but not including) 23 -- Since there was nothing left, stopping T200 counter -- Stopping T203 counter since we got an ACK -- Nothing left, starting T203 counter -- Got RR response to our frame -- Restarting T203 timer This is repeating every 10s or so... Any ideas what this message means and is there a way to prevent it from happening. No changes has been made on this asterisk box in years (running old 1.4.25 if it ain't broken version) Thanks in advance JY -- _ -- 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] Asterisk console suddenly extremely verbose...
Hi there. I started the console today to reload the extensions.conf file ; only to be greeted with extremely verbose console. Seems related to the zaptel card: Example: Supervisory frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] N(R): 020 P/F: 1 0 bytes of data voip*CLI [ 00 01 01 2f ] voip*CLI Supervisory frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] N(R): 023 P/F: 1 0 bytes of data Handling message for SAPI/TEI=0/0 -- ACKing all packets from 22 to (but not including) 23 -- Since there was nothing left, stopping T200 counter -- Stopping T203 counter since we got an ACK -- Nothing left, starting T203 counter -- Got RR response to our frame -- Restarting T203 timer This is repeating every 10s or so... Any ideas what this message means and is there a way to prevent it from happening. No changes has been made on this asterisk box in years (running old 1.4.25 if it ain't broken version) You have pri intense debug span x enabled. Disable with pri no debug span x. Richard -- _ -- 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] followme forking/parallel dialing breaks when 1 sip device unreachable
* Summary: I need to be able to ring multiple numbers in followme.conf at the same time, even if one of the SIP extensions is unreachable. This works in 1.4.8 but not in 1.8, just barfs and sends to voice mail instead of ringing the other 2 extensions on the same line in the followme.conf See more details below. * I decided to mess around with followme and it actually suits my needs quit well. I want to know what number the caller called into when my cell phone rings, then decide if I want to answer it by pressing 1 or not. Also helps with making sure voicemail is only left on my Asterisk voicemail instead of my cell phone voice mail. So, I set up followme on Asterisk 1.4.8 something like this and it worked great: from my followme.conf: number=2072065554441212,28 Problem is after moving this same config to my new 1.8 box the call fails and goes to voice mail if either of the two sip extensions are unreachable. So, let me explain further... If both SIP/207 and SIP/206 are up and running and accessible to receive the call then all goes well, if one of them is down for some reason then none of the 3 extensions ring and it just goes to voice mail. This stinks because I lose all calls to voice mail if for example my Internet connection goes down at home (207). Wouldn't this be the time you really want your other phones to ring? I thought about doing something like this: number=2072065554441212,28 number=2075554441212,28 In case say 206 fails but when 206 is up, they will be on hold for almost a minute before going to voice mail if I don't answer. I know there are other solutions outside followme but this worked in 1.4.8 and I have to think it should work in 1.8. Just not sure what I am missing. Thanks in advance for any help. --Todd -- _ -- 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] followme forking/parallel dialing breaks when 1 sip device unreachable
* Summary: I need to be able to ring multiple numbers in followme.conf at the same time, even if one of the SIP extensions is unreachable. This works in 1.4.8 but not in 1.8, just barfs and sends to voice mail instead of ringing the other 2 extensions on the same line in the followme.conf See more details below. * I decided to mess around with followme and it actually suits my needs quit well. I want to know what number the caller called into when my cell phone rings, then decide if I want to answer it by pressing 1 or not. Also helps with making sure voicemail is only left on my Asterisk voicemail instead of my cell phone voice mail. So, I set up followme on Asterisk 1.4.8 something like this and it worked great: from my followme.conf: number=2072065554441212,28 Problem is after moving this same config to my new 1.8 box the call fails and goes to voice mail if either of the two sip extensions are unreachable. So, let me explain further... If both SIP/207 and SIP/206 are up and running and accessible to receive the call then all goes well, if one of them is down for some reason then none of the 3 extensions ring and it just goes to voice mail. This stinks because I lose all calls to voice mail if for example my Internet connection goes down at home (207). Wouldn't this be the time you really want your other phones to ring? I thought about doing something like this: number=2072065554441212,28 number=2075554441212,28 In case say 206 fails but when 206 is up, they will be on hold for almost a minute before going to voice mail if I don't answer. I know there are other solutions outside followme but this worked in 1.4.8 and I have to think it should work in 1.8. Just not sure what I am missing. Thanks in advance for any help. This may work now after I fixed this issue last week on SVN v1.8: https://issues.asterisk.org/jira/browse/ASTERISK-17557 Do you get Extension '%s@%s' doesn't exist\n error messages? Richard -- _ -- 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] followme forking/parallel dialing breaks when 1 sip device unreachable
No, I get no error in the CLI at all, just shows that the followme is being executed then dumps straight to Vmail which is defined in my dialplan on the next line after calling the followme. I checked out the link and it also shows problems with callerid not passing, this is also a problem for me and that was what I was going to tackle next. I will checkout the patch, I have never applied a patch though, only done fresh installs. So, I will need to figure that out. I am running Asterisk 1.8.7.1 to be more specific. Thanks Richard. On Thu, Dec 15, 2011 at 8:56 PM, Richard Mudgett rmudg...@digium.comwrote: * Summary: I need to be able to ring multiple numbers in followme.conf at the same time, even if one of the SIP extensions is unreachable. This works in 1.4.8 but not in 1.8, just barfs and sends to voice mail instead of ringing the other 2 extensions on the same line in the followme.conf See more details below. * I decided to mess around with followme and it actually suits my needs quit well. I want to know what number the caller called into when my cell phone rings, then decide if I want to answer it by pressing 1 or not. Also helps with making sure voicemail is only left on my Asterisk voicemail instead of my cell phone voice mail. So, I set up followme on Asterisk 1.4.8 something like this and it worked great: from my followme.conf: number=2072065554441212,28 Problem is after moving this same config to my new 1.8 box the call fails and goes to voice mail if either of the two sip extensions are unreachable. So, let me explain further... If both SIP/207 and SIP/206 are up and running and accessible to receive the call then all goes well, if one of them is down for some reason then none of the 3 extensions ring and it just goes to voice mail. This stinks because I lose all calls to voice mail if for example my Internet connection goes down at home (207). Wouldn't this be the time you really want your other phones to ring? I thought about doing something like this: number=2072065554441212,28 number=2075554441212,28 In case say 206 fails but when 206 is up, they will be on hold for almost a minute before going to voice mail if I don't answer. I know there are other solutions outside followme but this worked in 1.4.8 and I have to think it should work in 1.8. Just not sure what I am missing. Thanks in advance for any help. This may work now after I fixed this issue last week on SVN v1.8: https://issues.asterisk.org/jira/browse/ASTERISK-17557 Do you get Extension '%s@%s' doesn't exist\n error messages? Richard -- _ -- 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] followme forking/parallel dialing breaks when 1 sip device unreachable
OK, read all about the patch, thanks for the fix Richard. I would like to apply this patch to my current 1.8.7.1 but I am afraid I don't have a clue how. Is this just a case of getting a copy of app_followme.c and replacing it on my current Asterisk install? If not, do I have to grab a new Asterisk version and recompile everything and start over? I know I can save my configs but I was hoping for a simple fix without having to recompile Asterisk from source etc. Thanks for any help. --Todd On Thu, Dec 15, 2011 at 9:07 PM, Todd Routhier fonema...@gmail.com wrote: No, I get no error in the CLI at all, just shows that the followme is being executed then dumps straight to Vmail which is defined in my dialplan on the next line after calling the followme. I checked out the link and it also shows problems with callerid not passing, this is also a problem for me and that was what I was going to tackle next. I will checkout the patch, I have never applied a patch though, only done fresh installs. So, I will need to figure that out. I am running Asterisk 1.8.7.1 to be more specific. Thanks Richard. On Thu, Dec 15, 2011 at 8:56 PM, Richard Mudgett rmudg...@digium.comwrote: * Summary: I need to be able to ring multiple numbers in followme.conf at the same time, even if one of the SIP extensions is unreachable. This works in 1.4.8 but not in 1.8, just barfs and sends to voice mail instead of ringing the other 2 extensions on the same line in the followme.conf See more details below. * I decided to mess around with followme and it actually suits my needs quit well. I want to know what number the caller called into when my cell phone rings, then decide if I want to answer it by pressing 1 or not. Also helps with making sure voicemail is only left on my Asterisk voicemail instead of my cell phone voice mail. So, I set up followme on Asterisk 1.4.8 something like this and it worked great: from my followme.conf: number=2072065554441212,28 Problem is after moving this same config to my new 1.8 box the call fails and goes to voice mail if either of the two sip extensions are unreachable. So, let me explain further... If both SIP/207 and SIP/206 are up and running and accessible to receive the call then all goes well, if one of them is down for some reason then none of the 3 extensions ring and it just goes to voice mail. This stinks because I lose all calls to voice mail if for example my Internet connection goes down at home (207). Wouldn't this be the time you really want your other phones to ring? I thought about doing something like this: number=2072065554441212,28 number=2075554441212,28 In case say 206 fails but when 206 is up, they will be on hold for almost a minute before going to voice mail if I don't answer. I know there are other solutions outside followme but this worked in 1.4.8 and I have to think it should work in 1.8. Just not sure what I am missing. Thanks in advance for any help. This may work now after I fixed this issue last week on SVN v1.8: https://issues.asterisk.org/jira/browse/ASTERISK-17557 Do you get Extension '%s@%s' doesn't exist\n error messages? Richard -- _ -- 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] Asterisk console suddenly extremely verbose...
Hi On 16 December 2011 13:24, Richard Mudgett rmudg...@digium.com wrote: You have pri intense debug span x enabled. Disable with pri no debug span x. Thanks... I couldn't find any configuration file showing this ; but ran the command in the CLI... Seems to have done it. I really wonder how it could have been turned on ... -- _ -- 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] Contexts and Extensions
Hello Everyone, For inbound, I am trying to specify a specific context. Everything works fine using the IP address, however with domain name it's not working at all. I tried changing the: Via: SIP/2.0/UDP test.com, and the Record-Route: sip:test.com;lr;did=a1a.4d23bae4 If I have a peer with the host, fromdomain, and outboundprxy set as the IP address the correct context is found context-from-test, but not using the domain name test.com. Asterisk still knows that the call is coming from IP address: chan_sip.c:22081 handle_request_invite: Call from '' (192.168.2.102:5060) to extension '1001' rejected because extension not found in context 'internal'. SIP Trace: --- SIP read from UDP:192.168.2.102:5060 --- INVITE sip:1...@test.com:5060 SIP/2.0 Record-Route: sip:test.com;lr;did=a1a.4d23bae4 Via: SIP/2.0/UDP test.com;branch=z9hG4bK9e83.1b0fcd74.0 Via: SIP/2.0/UDP 208.44.220.234:5060;received=208.44.220.234;branch=z9hG4bK6d6940f3;rport=5060 From: Mike Peer sip:16058293047@208.44.220.234;tag=as62765da7 To: sip:1001@170.12.90.130 Contact: sip:16058293047@208.44.220.234 Call-ID: 0f920dff6eefb6bd70b48d73676be593@208.44.220.234 CSeq: 102 INVITE User-Agent: DiDXsuPErTecSIP5 Max-Forwards: 69 Remote-Party-ID: Mike Peer sip:16058293047@208.44.220.234;privacy=off;screen=no Date: Fri, 16 Dec 2011 04:15:10 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 382 --- Reliably Transmitting (no NAT) to 192.168.2.102:5060 --- SIP/2.0 404 Not Found Via: SIP/2.0/UDP test.com;branch=z9hG4bK9e83.1b0fcd74.0;received=192.168.2.102 Via: SIP/2.0/UDP 208.44.220.234:5060;received=208.44.220.234;branch=z9hG4bK6d6940f3;rport=5060 From: Mike Peer sip:16058293047@208.44.220.234;tag=as62765da7 To: sip:1001@170.12.90.130;tag=as51f932b5 Call-ID: 0f920dff6eefb6bd70b48d73676be593@208.44.220.234 CSeq: 102 INVITE Server: Asterisk PBX UNKNOWN__and_probably_unsupported Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 I am using OpenSIPS and changed the following: advertised_address=test.com record_route_preset(test.com); Again, if I create a peer, and set the host, fromdomain, and outboundprxy as 192.168.2.102, and everything woks fine, but I would like to use the domain name example.com. Thanks in Advance, Nick. -- _ -- 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] Play audio file for both Caller and Callee in a call
Hi, Plese give a little example of script so that it will be clear. On Thu, Dec 15, 2011 at 11:09 PM, Jim Dickenson dicken...@cfmc.com wrote: You also use AMI to inject audio into the conversation using the ChanSpy application. -- Jim Dickenson mailto:dicken...@cfmc.com dicken...@cfmc.com CfMC http://www.cfmc.com/ On Dec 15, 2011, at 9:23 AM, Danny Nicholas wrote: You can’t per se, but you can call an AGI using stream? ** ** *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of * c.savinov...@itntelecom.com *Sent:* Thursday, December 15, 2011 11:22 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Play audio file for both Caller and Callee in a call ** ** Dear Danny: ** ** How can you use Playback in the middle of 2 channels engaged in a conversation? ** ** Thanks C. Savinovich ** ** Original Message Subject: Re: [asterisk-users] Play audio file for both Caller and Callee in a call From: Danny Nicholas da...@debsinc.com Date: Thu, December 15, 2011 9:31 am To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Playback? What flavor of Asterisk are you using? *From:* *asterisk-users-boun...@lists.digium.com*asterisk-users-boun...@lists.digium.com [*mailto:asterisk-users-boun...@lists.digium.com*asterisk-users-boun...@lists.digium.com ] *On Behalf Of *ISABEL ORDAS ARNAL *Sent:* Thursday, December 15, 2011 10:29 AM *To:* *asterisk-users@lists.digium.com* asterisk-users@lists.digium.com *Subject:* [asterisk-users] Play audio file for both Caller and Callee in a call Dear all, Anyone of you knows how to play an audio file at the beginning of a call for both Caller and Callee? A(x) of Dial application only plays audio for callee. I don’t want to use MeetMe because I want to use Monitor and MixMonitor. Thank you! -- Este mensaje se dirige exclusivamente a su destinatario. Puede consultar nuestra política de envío y recepción de correo electrónico en el enlace situado más abajo. This message is intended exclusively for its addressee. We only send and receive email on the basis of the terms set out at. *http://www.tid.es/ES/PAGINAS/disclaimer.aspx*http://www.tid.es/ES/PAGINAS/disclaimer.aspx -- -- _ -- Bandwidth and Colocation Provided by *http://www.api-digital.com*http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: *http://www.asterisk.org/hello* http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: *http://lists.digium.com/mailman/listinfo/asterisk-users*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 -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer -- _ -- 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] Play audio file for both Caller and Callee in a call
Use an AMI packet like this:
Action: Originate
Channel: Local/do_playback@cfmc_cdi_private
Exten: do_chanspy
Context: cfmc_cdi_private
Priority: 1
Variable: CfMC_ActionID=PlayBack
Variable: CfMC_WhatToPlay=lyrics-louie-louie
Variable: CfMC_WhoHear=SIP/GXP280
ActionID: PlayBack
Async: true
With dialplan like this:
exten = do_playback,1,Answer()
exten = do_playback,n,UserEvent(BeforePlayBack,ActionID:${CfMC_ActionID}
${UNIQUEID} ${CHANNEL} ${CfMC_WhatToPlay} ${CfMC_WhoHear})
exten = do_playback,n,Wait(0.3)
exten = do_playback,n,Playback(${CfMC_WhatToPlay})
; PLAYBACKSTATUS - SUCCESS FAILED
exten = do_playback,n,UserEvent(AfterPlayBack,ActionID:${CfMC_ActionID}
${UNIQUEID} ${CHANNEL} ${CfMC_WhatToPlay} ${CfMC_WhoHear}
${PLAYBACKSTATUS})
exten = do_playback,n,Hangup()
exten = do_chanspy,1,Answer()
exten = do_chanspy,n,UserEvent(BeforeChanSpy,ActionID:${CfMC_ActionID}
${UNIQUEID} ${CHANNEL} ${CfMC_WhatToPlay} ${CfMC_WhoHear})
exten = do_chanspy,n,ChanSpy(${CfMC_WhoHear},qW)
exten = do_chanspy,n,UserEvent(AfterChanSpy,ActionID:${CfMC_ActionID}
${UNIQUEID} ${CHANNEL} ${CfMC_WhatToPlay} ${CfMC_WhoHear})
exten = do_chanspy,n,Hangup()
You need to issue an AMI packet for each leg of the call. Each leg will hear
the same audio feed offset by however long it takes the packets to be
processed. In general this is a few milliseconds and should not be a big deal.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Dec 15, 2011, at 10:27 PM, virendra bhati wrote:
Hi,
Plese give a little example of script so that it will be clear.
On Thu, Dec 15, 2011 at 11:09 PM, Jim Dickenson dicken...@cfmc.com wrote:
You also use AMI to inject audio into the conversation using the ChanSpy
application.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Dec 15, 2011, at 9:23 AM, Danny Nicholas wrote:
You can’t per se, but you can call an AGI using stream?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
c.savinov...@itntelecom.com
Sent: Thursday, December 15, 2011 11:22 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Play audio file for both Caller and Callee in
a call
Dear Danny:
How can you use Playback in the middle of 2 channels engaged in a
conversation?
Thanks
C. Savinovich
Original Message
Subject: Re: [asterisk-users] Play audio file for both Caller and
Callee in a call
From: Danny Nicholas da...@debsinc.com
Date: Thu, December 15, 2011 9:31 am
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Playback? What flavor of Asterisk are you using?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of ISABEL ORDAS
ARNAL
Sent: Thursday, December 15, 2011 10:29 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Play audio file for both Caller and Callee in a
call
Dear all,
Anyone of you knows how to play an audio file at the beginning of a call for
both Caller and Callee?
A(x) of Dial application only plays audio for callee. I don’t want to use
MeetMe because I want to use Monitor and MixMonitor.
Thank you!
Este mensaje se dirige exclusivamente a su destinatario. Puede consultar
nuestra política de envío y recepción de correo electrónico en el enlace
situado más abajo.
This message is intended exclusively for its addressee. We only send and
receive email on the basis of the terms set out at.
http://www.tid.es/ES/PAGINAS/disclaimer.aspx
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_
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New to Asterisk? Join us for a live introductory webinar every Thurs:
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asterisk-users mailing list
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_
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asterisk-users mailing list
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asterisk-users mailing list
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Thanks and regards
Virendra Bhati
+91-8885268942
Software Engineer
--
