Re: [asterisk-users] Use of AGISIGHUP
Thanks for the replies. I am already ignoring the signal but it doesn't
seem to be making much difference on this system with this script. It's
an old legacy script I should hopefully be dropping and writing properly
within the dial plan.
I will keep trying!
Thanks
Lee
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve
Edwards
Sent: 26 August 2010 21:02
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Use of AGISIGHUP
On Thu, 26 Aug 2010, Lee Archer wrote:
I am setting AGISIGHUP=no before running a Perl script via AGI but
it doesn?t seem to be doing anything as the script is still exiting
on a hangup and not completing properly.
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve
Edwards
I'm just a 1.2 Luddite, so I've never seen AGISIGHUP and I think it's
a bad idea to protect lazy programmers :)
On Thu, 26 Aug 2010, Danny Nicholas wrote:
Here's a one-liner that should fix the problem
local $SIG{HUP} = 'IGNORE';
Does that make me lazy?
Not at all. If that is the correct response for your program, it's
perfect:
1) The program will execute correctly on your system, my system, any
system regardless of the configuration.
2) It tells the next guy explicitly what you intended to happen upon
receiving the signal.
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867
PST
Newline Fax:
+1-760-731-3000
--
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[asterisk-users] dynamic MeetMe, min. digits
Hi All, Is there a way to use the dynamic feature of the meetme application (D) and to set an option to configure the minimum length of the numbers for the conference and the associated pin. In my case, I'd like them to be at least four digits. Thanks in advance ! -- _ -- 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] dynamic MeetMe, min. digits
Xavier wrote: Hi All, Is there a way to use the dynamic feature of the meetme application (D) and to set an option to configure the minimum length of the numbers for the conference and the associated pin. You can use the read application to get the password and then check the length, before going onto the conference setup. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- 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] dynamic MeetMe, min. digits
Yes but what about the conference number ? On 08/27/2010 11:58 AM, Doug Lytle wrote: Xavier wrote: Hi All, Is there a way to use the dynamic feature of the meetme application (D) and to set an option to configure the minimum length of the numbers for the conference and the associated pin. You can use the read application to get the password and then check the length, before going onto the conference setup. Doug -- _ -- 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] music on hold in blind transfer
Hello, Is it possible to avoid playing music on hold during a blind transfer ? Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] queue agent and blind transfer
Hello, When an agent does a blind transfer the call hangups for him but shows as In use in queue in my CRM (used for auto dialing). As a result the agent have to wait until the transfered call completes. Is there any way to change this behaviour ? -- _ -- 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 Forwarding
Hi, I'm currently programming an interface for my Asterisk service. I've noticed an issue if someone sets up call forwarding on their sip phone. Asterisk receives a 302 Moved Temporarily message, and forwards the call successfully. However, the CDR is not correct. If I set up call forwarding to a mobile on extension 201, and then place a call from extension 202, the call gets diverted. I answer the call and talk for 30 seconds, then I hang up. The CDR shows two calls:- 2010-08-27 13:38:24 - 202 - 201 - Answered - Billsec is 30 2010-08-27 13:38:24 - 202 - 5551234 - Answered - Billsec is 0 5551234 is the mobile number. The second CDR entry should read 30 seconds, and the first should read 0 (or 30) Since it isn't behaving like I want, is there any way to disable the feature that allows a SIP phone to perform call forwarding? Thanks Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] music on hold in blind transfer
On Fri, Aug 27, 2010 at 7:39 AM, Tino t...@sparksupport.com wrote: Is it possible to avoid playing music on hold during a blind transfer ? Please do not cross-post the same message to multiple lists. Yes, configure an empty MoH class or not loading MoH are some options, also: *CLI core show application Dial -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Duplicate channel variables after transfer
Hi all, with an (attended) transfer i see the following happening: 1) A calls B1 2) B2 calls C 3) B2 transfers call to A 4) A talks to C At step 3, the channel A is connected to channel C and B1 and B2 are hung up. In the h extension for channel B2, the channel is renamed to B2ZOMBIE and i see that the channel variables of A have been merged into B2ZOMBIE. If there were duplicate names for variables, the channel now has those variables doubled. The DumpChan() application called from the h extension confirms this. In my case the channels are all SIP channels and in the h extension I want to access the SIPCALLID variable of the A channel. Every access to it gives me the wrong value namely that of channel B1. How do i access the _second_ variable named SIPCALLID in the channel? Extract from DumpChan() as an example: Dumping Info For Channel: SIP/sipout-0055ZOMBIE: Info: Name= SIP/sipout-0055ZOMBIE Type= SIP UniqueID= 1282913436.108 Variables: ... sipcallid=eae94252-ebf23...@172.28.4.112 ... sipcallid=lyvkqtybsgrt...@172.28.4.113 ... I want to get lyvkqtybsgrt...@172.28.4.113 instead of eae94252- ebf23...@172.28.4.112 as a result. -- Greetings, Alex Hermann -- _ -- 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 Forwarding
Dan Journo schrieb: Since it isn't behaving like I want, is there any way to disable the feature that allows a SIP phone to perform call forwarding? Thanks Dan Hello, in asterisk 1.6.x there is a Dial option i which suppress a 302 redirect which is very nice when dialing more than one phone at once, but you can use it also if you just dial one channel. see output of core show application dial: i- Asterisk will ignore any forwarding requests it may receive on this dial attempt. best regards steve -- Für weitere Fragen stehen wir gerne unter v...@sil.at oder 059944 - 2440 zur Verfügung. Mit freundlichen Grüssen -- Stefan Schmidt Sysadmin/VOIP // v...@sil.at // Tel 059944-2440// - SILVER SERVER GmbH // Lorenz-Mandl-Gasse 33/1 // A-1160 Wien // Fax 059944-9000 // www.sil.at // - -- _ -- 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] CDR on Transfer...
Did you find the solution? On Thu, Aug 26, 2010 at 7:25 PM, Carlos Chavez cur...@telecomabmex.comwrote: I have searched for some time but I have not found an asnwer on how to fix the CDR when a call is transferred. The problem is that if someone dials a cell phone and then transfers the call to another extensión the CDR for the cell call stops and there is no way to track that the call was transferred so we can bill correctly. Many people have asked this question but there is no answer, only a mention that it should be fixed in 1.6 which it is not (at least on 1.6.2.11). Any tips oh how to correct this problem? A lot of customers give me grief about this because they cannot properly bill people for their cell calls. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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-users Digest, Vol 73, Issue 58
: [asterisk-users] Use of AGISIGHUP
On Thu, 26 Aug 2010, Lee Archer wrote:
I am setting AGISIGHUP=no before running a Perl script via AGI but
it doesn?t seem to be doing anything as the script is still exiting
on a hangup and not completing properly.
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve
Edwards
I'm just a 1.2 Luddite, so I've never seen AGISIGHUP and I think it's
a bad idea to protect lazy programmers :)
On Thu, 26 Aug 2010, Danny Nicholas wrote:
Here's a one-liner that should fix the problem
local $SIG{HUP} = 'IGNORE';
Does that make me lazy?
Not at all. If that is the correct response for your program, it's
perfect:
1) The program will execute correctly on your system, my system, any
system regardless of the configuration.
2) It tells the next guy explicitly what you intended to happen upon
receiving the signal.
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867
PST
Newline Fax:
+1-760-731-3000
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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--
Message: 13
Date: Fri, 27 Aug 2010 11:27:57 +0200
From: Xavier magicrhe...@ouranos.be
Subject: [asterisk-users] dynamic MeetMe, min. digits
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID: 4c77851d.4090...@ouranos.be
Content-Type: text/plain; charset=iso-8859-1
Hi All,
Is there a way to use the dynamic feature of the meetme application (D)
and to set an option to configure the minimum length of the numbers for
the conference and the associated pin.
In my case, I'd like them to be at least four digits.
Thanks in advance !
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Message: 14
Date: Fri, 27 Aug 2010 05:58:57 -0400
From: Doug Lytle supp...@drdos.info
Subject: Re: [asterisk-users] dynamic MeetMe, min. digits
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID: 4c778c61.4080...@drdos.info
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Xavier wrote:
Hi All,
Is there a way to use the dynamic feature of the meetme application
(D) and to set an option to configure the minimum length of the
numbers for the conference and the associated pin.
You can use the read application to get the password and then check the
length, before going onto the conference setup.
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety.
--
Message: 15
Date: Fri, 27 Aug 2010 12:28:38 +0200
From: Xavier D. magicrhe...@ouranos.be
Subject: Re: [asterisk-users] dynamic MeetMe, min. digits
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID: 4c779356.1070...@ouranos.be
Content-Type: text/plain; charset=iso-8859-1
Yes but what about the conference number ?
On 08/27/2010 11:58 AM, Doug Lytle wrote:
Xavier wrote:
Hi All,
Is there a way to use the dynamic feature of the meetme application
(D) and to set an option to configure the minimum length of the
numbers for the conference and the associated pin.
You can use the read application to get the password and then check the
length, before going onto the conference setup.
Doug
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Message: 16
Date: Fri, 27 Aug 2010 17:09:33 +0530
From: Tino t...@sparksupport.com
Subject: [asterisk-users] music on hold in blind transfer
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID:
aanlktikfks7jcw-vkobvo6cbw0fc+-koc9ndhso1p...@mail.gmail.com
Content-Type: text/plain; charset=iso-8859-1
Hello,
Is it possible to avoid playing music on hold during a blind transfer ?
Thanks
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Message: 17
Date: Fri, 27 Aug 2010 17:35:26 +0530
From
Re: [asterisk-users] asterisk-users Digest, Vol 73, Issue 58
: [asterisk-users] Use of AGISIGHUP
On Thu, 26 Aug 2010, Lee Archer wrote:
I am setting AGISIGHUP=no before running a Perl script via AGI but
it doesn?t seem to be doing anything as the script is still exiting
on a hangup and not completing properly.
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve
Edwards
I'm just a 1.2 Luddite, so I've never seen AGISIGHUP and I think it's
a bad idea to protect lazy programmers :)
On Thu, 26 Aug 2010, Danny Nicholas wrote:
Here's a one-liner that should fix the problem
local $SIG{HUP} = 'IGNORE';
Does that make me lazy?
Not at all. If that is the correct response for your program, it's
perfect:
1) The program will execute correctly on your system, my system, any
system regardless of the configuration.
2) It tells the next guy explicitly what you intended to happen upon
receiving the signal.
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867
PST
Newline Fax:
+1-760-731-3000
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Message: 13
Date: Fri, 27 Aug 2010 11:27:57 +0200
From: Xavier magicrhe...@ouranos.be
Subject: [asterisk-users] dynamic MeetMe, min. digits
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID: 4c77851d.4090...@ouranos.be
Content-Type: text/plain; charset=iso-8859-1
Hi All,
Is there a way to use the dynamic feature of the meetme application (D)
and to set an option to configure the minimum length of the numbers for
the conference and the associated pin.
In my case, I'd like them to be at least four digits.
Thanks in advance !
-- next part --
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Message: 14
Date: Fri, 27 Aug 2010 05:58:57 -0400
From: Doug Lytle supp...@drdos.info
Subject: Re: [asterisk-users] dynamic MeetMe, min. digits
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID: 4c778c61.4080...@drdos.info
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Xavier wrote:
Hi All,
Is there a way to use the dynamic feature of the meetme application
(D) and to set an option to configure the minimum length of the
numbers for the conference and the associated pin.
You can use the read application to get the password and then check the
length, before going onto the conference setup.
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety.
--
Message: 15
Date: Fri, 27 Aug 2010 12:28:38 +0200
From: Xavier D. magicrhe...@ouranos.be
Subject: Re: [asterisk-users] dynamic MeetMe, min. digits
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID: 4c779356.1070...@ouranos.be
Content-Type: text/plain; charset=iso-8859-1
Yes but what about the conference number ?
On 08/27/2010 11:58 AM, Doug Lytle wrote:
Xavier wrote:
Hi All,
Is there a way to use the dynamic feature of the meetme application
(D) and to set an option to configure the minimum length of the
numbers for the conference and the associated pin.
You can use the read application to get the password and then check the
length, before going onto the conference setup.
Doug
-- next part --
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Message: 16
Date: Fri, 27 Aug 2010 17:09:33 +0530
From: Tino t...@sparksupport.com
Subject: [asterisk-users] music on hold in blind transfer
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID:
aanlktikfks7jcw-vkobvo6cbw0fc+-koc9ndhso1p...@mail.gmail.com
Content-Type: text/plain; charset=iso-8859-1
Hello,
Is it possible to avoid playing music on hold during a blind transfer ?
Thanks
-- next part --
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Message: 17
Date: Fri, 27 Aug 2010 17:35:26 +0530
From
Re: [asterisk-users] Call Forwarding
in asterisk 1.6.x there is a Dial option Sorry, any solutions for Asterisk 1.4? Thanks Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk DTMF RFC2833 issues
Hi all I have posted a question on the asterisk dev board about this issue but I want to see if any users have run up against this. This issue is that when calls are run through Broadvox and Level 3 the in-call rfc2833 dtmf is not reliable. This occured for me on asterisk version 1.6.1.18, 1.6.1.20 it appears to have been fixed when I went to 1.6.2.11 but broken again in 1.6.2.12-rc1. I have tested with Grandstream and SNOM phones and both fail 90% of the time Unidata phones fail 10% of the time Audiocodes and Grandstream ATA's appear to not suffer from the issue on any version of asterisk. What happens is when a caller trys to enter DTMF keys durring a call the far end routed through these carriers do not detect all of the digits. We did captures with broadvox and here is what they have said. Hello, Per our phone conversation I have attached our signaling capture. The issue is that after we receive a RTP packet, the RTP event that follows needs to be sent within 100 ms. Anything greater than 100 ms will not be received. Thank you, Broadvox Network Operations Center Any one else seen this? Any ideas? Please note you must be being proxied directly to the carrier so your RTP flows direct other wise you will not see the issue. Thanks Bryant -- _ -- 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] Protect yourself
Hey all We are seeing intrusion attempts coming from address 201.47.236.122 today They were hitting our switches trying to get in. So we blocked them at our firewall. Just wanted to put the word out so you all can protect your self. Bryant -- _ -- 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 CDR file Master.csv
How can we set the CDR Master file to rollover at say 30 Meg and create a new one -- _ -- 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] TELUS British Columbia PRI Settings
I am having some difficulties getting my Asterisk box to find the d-channel from a TELUS PRI and am waiting to hear back from one of their techs. In the meantime I thought I would check with the brilliant people of the mailing list. As I understand it is a T1 connection, not an E1 and I am using a Digium TE121 with hardware echo cancellation. I have green lights on the back of the card and the PRI connection, which go red when I do a dahdi restart command and come back to green once it is finished. I know from our other system that the frame and coding are ESF and B8ZS so it must be something in the signaling of the channels. My chan_dadhdi is below, the commented out lines are ones I have tried. I’ve also tried moving the dchannel around, 12 through 24. Does anyone see anything blatantly wrong? Thanks, Jeremy chan_dahdi.conf [trunkgroups] trunkgroup = 1,24 spanmap = 1,1,0,esf,b8zs [channels] #include /etc/asterisk/dahdi-channels.conf ;signalling = em ;signalling = pri_net signalling = pri_cpe ;context=default context=incoming ;switchtype=national switchtype=dms100 ;group = 1 bchannel = 1-12 dchannel = 24 echocancel=yes -- _ -- 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 CDR file Master.csv
- Ujjval Karihaloo ujj...@simplesignal.com wrote: How can we set the CDR Master file to rollover at say 30 Meg and create a new one Use 'logrotate'. --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ASterisk CDR file Master.csv
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tim Nelson Subject: Re: [asterisk-users] ASterisk CDR file Master.csv - Ujjval Karihaloo ujj...@simplesignal.com wrote: How can we set the CDR Master file to rollover at say 30 Meg and create a new one Use 'logrotate'. --Tim To improve on your answer, set up a shell to check the size of CDR master and do a logrotate (/usr/sbin/asterisk -rx logger rotate) when the condition is met. GIYF on this one. -- _ -- 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 CDR file Master.csv
On 8/27/2010 11:55 AM, Danny Nicholas wrote: *From:* asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Tim Nelson *Subject:* Re: [asterisk-users] ASterisk CDR file Master.csv - Ujjval Karihaloo ujj...@simplesignal.com wrote: How can we set the CDR Master file to rollover at say 30 Meg and create a new one Use 'logrotate'. --Tim To “improve” on your answer, set up a shell to check the size of CDR master and do a logrotate (/usr/sbin/asterisk –rx “logger rotate”) when the condition is met. GIYF on this one. I use logrotate to help with the files in /var/log/asterisk, but it does nothing with Master.csv. I am working on a script described earlier that if the file gets larger than a certain number of lines to move them off to another file and compress for space. If that is something anyone else is interested in, let me know and I'll post it when it's working. -- Dean Hoover Milwaukee, Wisconsin -- _ -- 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] TELUS British Columbia PRI Settings
On Fri, Aug 27, 2010 at 12:46 PM, jeremy.hellst...@synovate.com wrote: moving the dchannel around, 12 through 24. Does anyone see anything blatantly wrong? What alarms are you getting? -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.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] TELUS British Columbia PRI Settings
I see ... Chan_dahdi.c 2796 pri_find_dchan No D-channel available using Primary channel X as D-channel anyway. With X being whichever number I assigned to the D-channel in chan_dahdi and system.conf. Then when dialling I get an error 0 - unknown, which occurs when Asterisk tries to open a channel from softphone to PSTN. Jeremy -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paul Belanger Sent: August 27, 2010 10:13 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] TELUS British Columbia PRI Settings On Fri, Aug 27, 2010 at 12:46 PM, jeremy.hellst...@synovate.com wrote: moving the dchannel around, 12 through 24. Does anyone see anything blatantly wrong? What alarms are you getting? -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.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 -- _ -- 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 CDR file Master.csv
Thx Dean. I will be interested in testing that as well. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dean Hoover Sent: Friday, August 27, 2010 11:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] ASterisk CDR file Master.csv On 8/27/2010 11:55 AM, Danny Nicholas wrote: *From:* asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Tim Nelson *Subject:* Re: [asterisk-users] ASterisk CDR file Master.csv - Ujjval Karihaloo ujj...@simplesignal.com wrote: How can we set the CDR Master file to rollover at say 30 Meg and create a new one Use 'logrotate'. --Tim To improve on your answer, set up a shell to check the size of CDR master and do a logrotate (/usr/sbin/asterisk -rx logger rotate) when the condition is met. GIYF on this one. I use logrotate to help with the files in /var/log/asterisk, but it does nothing with Master.csv. I am working on a script described earlier that if the file gets larger than a certain number of lines to move them off to another file and compress for space. If that is something anyone else is interested in, let me know and I'll post it when it's working. -- Dean Hoover Milwaukee, Wisconsin -- _ -- 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] GXP-2000 transfer hold problem
Hi all, I'm working on a system with 4 Grandstream GP-200 Phones and the base Asterisk install. I have added a 5 phone which is remote to the client and located in my office. I can't get the phone to transfer a call or put a call on hold. This applies to all the phones at the location. I have been looking over configs and I'm at a loss right now. Any help in pointing this out would be greatly 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
[asterisk-users] Asterisk Crashed - But why?
Asterisk crashes from time to time and dumps core. So... what do I do with it? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Crashed - But why?
What do the logs in /var/log/asterisk/* tell you? Tim Nelson Systems/Network Support Rockbochs Inc. (218)727-4332 x105 - Jayson Baker jay...@spectrasurf.com wrote: Asterisk crashes from time to time and dumps core. So... what do I do with it? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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 Crashed - But why?
On Fri, 27 Aug 2010, Jayson Baker wrote: Asterisk crashes from time to time and dumps core. So... what do I do with it? Depending on the version, start reading asterisk-source-directory/doc/README.backtrace or asterisk-source-directory/doc/backtrace.txt. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR on Transfer...
Carlos Chavez cur...@telecomabmex.com writes: I have searched for some time but I have not found an asnwer on how to fix the CDR when a call is transferred. The problem is that if someone dials a cell phone and then transfers the call to another extensión the CDR for the cell call stops and there is no way to track that the call was transferred so we can bill correctly. Many people have asked this question but there is no answer, only a mention that it should be fixed in 1.6 which it is not (at least on 1.6.2.11). You can set a TRANSFERCONTEXT. In that context you can try to use ForkCDR and its companions to get the records right. If you come up with a setup which acts perfectly no matter the scenario I would be happy to hear about it. Note that TRANSFERCONTEXT is not invoked when the phone does a SIP redirect before the call is answered, AFAIK. Notice that it's been a long time since I battled with this part of Asterisk, and I didn't check that I remembered correctly. This will all be a lot more sane with Channel Event Logging in 1.8.x, but at that point you need to run mediation before you get CDR's you can use for billing. /Benny -- _ -- 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] CDR on Transfer...
to get accurate cdr's i just use a border server to send every call through that logs cdr... doesn't matter how many times it gets transferred internally the border server still gets accurate records of the whole call. On 27 August 2010 21:07, Benny Amorsen benny+use...@amorsen.dkbenny%2buse...@amorsen.dk wrote: Carlos Chavez cur...@telecomabmex.com writes: I have searched for some time but I have not found an asnwer on how to fix the CDR when a call is transferred. The problem is that if someone dials a cell phone and then transfers the call to another extensión the CDR for the cell call stops and there is no way to track that the call was transferred so we can bill correctly. Many people have asked this question but there is no answer, only a mention that it should be fixed in 1.6 which it is not (at least on 1.6.2.11). You can set a TRANSFERCONTEXT. In that context you can try to use ForkCDR and its companions to get the records right. If you come up with a setup which acts perfectly no matter the scenario I would be happy to hear about it. Note that TRANSFERCONTEXT is not invoked when the phone does a SIP redirect before the call is answered, AFAIK. Notice that it's been a long time since I battled with this part of Asterisk, and I didn't check that I remembered correctly. This will all be a lot more sane with Channel Event Logging in 1.8.x, but at that point you need to run mediation before you get CDR's you can use for billing. /Benny -- _ -- 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] CDR on Transfer...
Please don't top-post. Geraint Lee gera...@gmail.com writes: to get accurate cdr's i just use a border server to send every call through that logs cdr... doesn't matter how many times it gets transferred internally the border server still gets accurate records of the whole call. That is what we do too, but customers are requesting CDR's which include information about e.g. which specific phone answered the call. This information is unknown to the border servers. We provide customers with access to the CDR's generated on their particular virtual Asterisk, but we receive complaints about the deficiencies of the 1.6.x CDR's. It is particularly troublesome that dial plan changes often change CDR's. With Channel Event Logging we should be able to provide all the information which customers ask for and at the same time insulate them from dial plan changes by logging only the information we want in precisely the format we wnat. I look forward to that, even though it means a bit of work mediating the logs before presenting them to customers. /Benny -- _ -- 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 Crashed - But why?
There is nothing in /var/log/asterisk... hmm, which log should I turn on? Debug? On Fri, Aug 27, 2010 at 1:25 PM, Tim Nelson tnel...@rockbochs.com wrote: What do the logs in /var/log/asterisk/* tell you? Tim Nelson Systems/Network Support Rockbochs Inc. (218)727-4332 x105 - Jayson Baker jay...@spectrasurf.com wrote: Asterisk crashes from time to time and dumps core. So... what do I do with it? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] Migrating 1.4 to 1.6.2
much static testing of my realtime configuration and applications I'm almost ready to pull the trigger. The one thing I've been able to determine is what I need to do to migrate my g729 licenses. Has anyone got any advice for me on this? The Digium site is... difficult to navigate TIA Bruce Ferrell -- _ -- 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] Compiling snmp_res.so into AsteriskNow install
I installed AsteriskNow1.7 and am trying to load the res_snmp module to monitor the system. Am I correct in saying that compiling asterisk from source and including the module is the only way to accomplish this? I’m a little worried about simply downloading the same source version as my current build (asterisk 1.6.1.2.11-2) and running a make menuselect? Thanks! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Migrating 1.4 to 1.6.2
From: Bruce Ferrell bferr...@baywinds.org much static testing of my realtime configuration and applications I'm almost ready to pull the trigger. The one thing I've been able to determine is what I need to do to migrate my g729 licenses. Has anyone got any advice for me on this? The Digium site is... difficult to navigate TIA Bruce Ferrell--- If you are not changing servers you just download the correct binary for 1.6.2 for your machine. If your are moving machines then you must re-register the license on the new box. If you have moved them before you must call Digium and have them increment the count on the licenses. Here is a link to the general install instructions. http://downloads.digium.com/pub/telephony/codec_g729/README It is not really hard to do you just need to follow the steps. Bryant -- _ -- 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] Early media and IAX2
My IAX2 trunk provider, Teliax, seems to be forcing early media. Early media is cool and all, but my Asterisk install doesn't seem to be fully supporting it. My initial setting was using Dial() to call all of my dahdi (TDM400P) extensions. The results were that incoming calls would not hear any ringing tones and the call would be ended by Teliax after 21 seconds. Looking at the packet dumps, my asterisk box is sending an ACCEPT and a RINGING packet to Teliax. I tried: Progress(); Dial(exten); Progress(); Ringing(); Dial(exten); Ringing(); Dial(exten); Dial(exten,,r); Progress(); Dial(exten,,r); All with the same result. After some experimenting, I found that: Playback(tt-weasels,noanswer); Would playback tt-weasels to the caller in early media. Furthermore: Dial(exten,,m); Plays back moh to the caller and also allows the ringing portion of the call to pass the 21 second mark. If I had a ringing moh mp3, it would be everything I wanted. However, it seems like the wrong solution. Not only should Dial(exten,,r); work, but it seems like Dial(exten); should know to send the ringing tones. My asterisk version is 1:1.6.2.9-1 on debian/sid. -- _ -- 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 1.6 Displaying BackGround() in call trace but no audio is heard from caller
Thought a different succinct subject line must drum up an answer or two...
Also, this has been tested from two different carriers: We're getting
an average of 2/10 call success rate.
-- Forwarded message --
From: Joe Wood sch...@gmail.com
Date: Thu, Aug 26, 2010 at 6:58 PM
Subject: Asterisk 1.6 Displaying in Debug that it is playing a ulaw
file using BackGround() but no audio is heard from the phone
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
First off, let me first say that this is not a one-way audio problem.
Sometimes I can get 'her' to speak to me, other times I can't for a
long time.
I'm just using a very simple system to dump people into MeetMe().
Nothing fancy.
I do have the following in my modules.conf:
preload = format_mp3.so
preload = codec_ulaw.so
preload = format_pcm.so
My extensions.conf looks like:
[general]
autofallthrough=yes
static=no
writeprotect=no
extenpatternmatchnew=yes
clearglobalvars=no
[conference-calls]
exten = s,1,Answer()
exten = s,n,Background(welcome)
exten = s,n,Background(and)
exten = s,n,Background(thank-you-for-calling)
exten = s,n,Background(conference-reservations)
exten = s,n,Wait(2)
exten = s,n,Background(enter-conf-pin-number)
exten = s,n,WaitExten(10)
exten = i,1,Playback(pbx-invalid)
exten = i,n,Goto(conference-calls,9000,1)
exten = t,1,Playback(vm-goodbye)
exten = t,n,Hangup()
exten = ${EXTEN},1,Meetme(${EXTEN})
== Using SIP RTP CoS mark 5
-- Executing [...@conference-calls:1]
Answer(SIP/2063161626-0001, ) in new stack
== Using SIP RTP CoS mark 5
-- Executing [...@conference-calls:1]
Answer(SIP/2063161626-0002, ) in new stack
-- Executing [...@conference-calls:2]
BackGround(SIP/2063161626-0001, welcome) in new stack
-- SIP/2063161626-0001 Playing 'welcome.ulaw' (language 'en')
-- Executing [...@conference-calls:2]
BackGround(SIP/2063161626-0002, welcome) in new stack
-- SIP/2063161626-0002 Playing 'welcome.ulaw' (language 'en')
-- Executing [...@conference-calls:3]
BackGround(SIP/2063161626-0001, and) in new stack
-- SIP/2063161626-0001 Playing 'and.ulaw' (language 'en')
-- Executing [...@conference-calls:3]
BackGround(SIP/2063161626-0002, and) in new stack
-- SIP/2063161626-0002 Playing 'and.ulaw' (language 'en')
-- Executing [...@conference-calls:4]
BackGround(SIP/2063161626-0001, thank-you-for-calling) in new
stack
-- SIP/2063161626-0001 Playing 'thank-you-for-calling.ulaw'
(language 'en')
-- Executing [...@conference-calls:4]
BackGround(SIP/2063161626-0002, thank-you-for-calling) in new
stack
-- SIP/2063161626-0002 Playing 'thank-you-for-calling.ulaw'
(language 'en')
-- Executing [...@conference-calls:5]
BackGround(SIP/2063161626-0001, conference-reservations) in
new stack
-- SIP/2063161626-0001 Playing
'conference-reservations.ulaw' (language 'en')
-- Executing [...@conference-calls:5]
BackGround(SIP/2063161626-0002, conference-reservations) in
new stack
-- SIP/2063161626-0002 Playing
'conference-reservations.ulaw' (language 'en')
-- Executing [...@conference-calls:6]
Wait(SIP/2063161626-0001, 2) in new stack
-- Executing [...@conference-calls:6]
Wait(SIP/2063161626-0002, 2) in new stack
-- Executing [...@conference-calls:7]
BackGround(SIP/2063161626-0001, enter-conf-pin-number) in new
stack
-- SIP/2063161626-0001 Playing 'enter-conf-pin-number.ulaw'
(language 'en')
-- Executing [...@conference-calls:7]
BackGround(SIP/2063161626-0002, enter-conf-pin-number) in new
stack
-- SIP/2063161626-0002 Playing 'enter-conf-pin-number.ulaw'
(language 'en')
-- Executing [...@conference-calls:8]
WaitExten(SIP/2063161626-0001, 10) in new stack
-- Executing [...@conference-calls:8]
WaitExten(SIP/2063161626-0002, 10) in new stack
-- Timeout on SIP/2063161626-0001, going to 't'
-- Executing [...@conference-calls:1]
Playback(SIP/2063161626-0001, vm-goodbye) in new stack
-- SIP/2063161626-0001 Playing 'vm-goodbye.ulaw' (language 'en')
-- Timeout on SIP/2063161626-0002, going to 't'
-- Executing [...@conference-calls:1]
Playback(SIP/2063161626-0002, vm-goodbye) in new stack
-- SIP/2063161626-0002 Playing 'vm-goodbye.ulaw' (language 'en')
-- Executing [...@conference-calls:2]
Hangup(SIP/2063161626-0001, ) in new stack
== Spawn extension (conference-calls, t, 2) exited non-zero on
'SIP/2063161626-0001'
-- Executing [...@conference-calls:2]
Hangup(SIP/2063161626-0002, ) in new stack
== Spawn extension (conference-calls, t, 2) exited non-zero on
'SIP/2063161626-0002'
Has anyone else encountered this problem before? I saw one posting on
the listserv, but it said to add in the pcm lib and I did that and no
change.
Help.
Thanks a bunch,
Joe
--
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Re: [asterisk-users] IAX2 and other modules load error on 1.8.0-beta3
I'm sorry, I tried this but the SVN version does not seem to work on my machine. I get no DAHDI support, I can't even select it in menuselect so I've no idea what to do. Ira At 11:28 AM 8/23/2010, you wrote: On Monday 23 August 2010 12:19:38 Ira wrote: At 09:26 AM 8/23/2010, you wrote: There's already an issue open for this, AND there is a patch posted, BUT the reporter needs to verify that the patch(es) fix the issue for him: https://issues.asterisk.org/view.php?id=17707 And how was I supposed to now that? I being the reporter. I hate to seem stupid, but when I got the email I looked there but have no idea what I'm supposed to do or how to do it. What is a patch and what do I do with it? In the future (and this goes for everybody, not just you), if you do not understand a request made by a developer, PLEASE ask that question, instead of giving us no feedback whatsoever. We're not trying to stump you or make you feel stupid, honest; we just need to verify that a proposed solution fixes the problem reported and not some other, unrelated problem. I'll defer to Paul's excellent set of instructions as to how to test a proposed patch. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) 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 -- _ -- 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] Youmail RDNIS
I don't see why it does not work. Setting RDNIS and calling most GSM mobile phones produces a forwarded call annoucement, so why would the do it any different? We get RDNIS in a SIP field and use it to keep the same voicemail for a desk phone and cell phone, also can forward ILEC and most CLEC remote call forwarding and get the correct info, or forward a cell phone to RCF to DID and I see the entire route. If you have a T1 with many DIDs and a provider that supports it you can have all DID forward to a single DID elsewhere and still be able to route by dialed number. All if this done with RDNIS. Are you sure your provider is consistantly sending it? On Wed, Aug 11, 2010 at 17:54, Karl Fife karlf...@gmail.com wrote: into the voicemail account belonging to the RDNIS value. In practice I find that YouMail, when presented with a redirected call as described above ignores the RDNIS value and prompts me for an subscriber account number. By contrast, when various MNO's do the redirection, YouMail is able to determine the redirecting subscriber account number--presumably some other way. Does anyone know the mechanism(s) by which this is normally done? Is there even a 'normal' way to do this (such as a QSIG call transfer message), or is truly home-spun and carrier-specific, such as a Q.931 facility message. Any advice on the subject would be much appreciated! Thanks! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] outbound SIP trunk hunting (or any fxo for that matter)
Hi! My question is this. Is it possible to tell Asterisk to execute part of a macro as a block without allowing any other commands to be processed during that time? What would be a correct way to do this in 1.4.x? *CLI show application MacroExclusive Philipp -- _ -- 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
