Re: [asterisk-users] Trouble with *8 Pickup
On Thu, 2011-08-11 at 16:38 +0100, Paul Hayes wrote: 2011/8/11 Ishfaq Malik i...@pack-net.co.uk mailto:i...@pack-net.co.uk On Thu, 2011-08-11 at 14:47 +0100, --[ UxBoD ]-- wrote: Ah, now this is interesting as one of our clients had the same problem the other day; in our case when they performed the *8 they got an extension unavailable from a completely different dialplan! This was on Asterisk 1.6 though with Snom phones. In the case of this server I was looking at, the only time this error occurred was when the pickup request happened in the same second as a dialplan step change so by the time the pick up of the channel was attempted, it no longer existed. -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 It's not just a snom/asterisk thing. I can replicate this with various phones and Asterisk 1.8.5. In fact with some phones the symptoms seemed worse where the phone *8 had been dialled on didn't hang up but thought it was on a call (while the caller had gone through to whatever the next dial plan priority was, a Queue in my test case). It makes perfect sense to me that a pickup should fail if your Dial has finished and * is stepping onto the next priority but a nicer Warning such as Trying to pickup a non-existent channel would be better. My test code was simply this: exten = 123321,1,Dial(SIP/5502,5) same = n,Answer same = n,Wait(1) same = n,Queue(booking,thHr) If you time the *8 just right so it is being handled during the end of the Dial then I got: [Aug 11 16:26:18] ERROR[18458]: astobj2.c:110 INTERNAL_OBJ: user_data is NULL [Aug 11 16:26:18] ERROR[18458]: astobj2.c:110 INTERNAL_OBJ: user_data is NULL [Aug 11 16:26:18] WARNING[18458]: chan_sip.c:6429 sip_fixup: No SIP tech_pvt! Fixup of SIP/5501-01da failed. [Aug 11 16:26:18] WARNING[18458]: channel.c:6462 ast_do_masquerade: Fixup failed on channel SIP/5501-01daMASQ, strange things may happen. cheers, Paul. Have you seen it in any other versions of 1.8 or is it something that has happened in the latest release? -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- 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] Trouble with *8 Pickup
On 12/08/11 08:46, Ishfaq Malik wrote: Have you seen it in any other versions of 1.8 or is it something that has happened in the latest release? I've not specifically seen this issue with other versions of Asterisk but then I've never tried to replicate it. The only time I've seen this with 1.8.5 is when I've purposely replicated it after reading your post. I have had much, much worse problems with pickup in previous versions of 1.8 and in the 1.6 branch where pickup will occasionally lock chan_sip altogether. This is a known issue and is in Jira and is fixed in 1.8.5. This issue doesn't really seem to cause any problems other than some stuck SIP channels. It's in Jira too: https://issues.asterisk.org/jira/browse/ASTERISK-18225 For the minute I can live with this and a nightly cron job to restart Asterisk to drop the stuck channels. Bit of a bodge I know but it works till someone fixes the issue. cheers, Paul. -- _ -- 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] Trouble with *8 Pickup
On Fri, 2011-08-12 at 09:46 +0100, Paul Hayes wrote: On 12/08/11 08:46, Ishfaq Malik wrote: Have you seen it in any other versions of 1.8 or is it something that has happened in the latest release? I've not specifically seen this issue with other versions of Asterisk but then I've never tried to replicate it. The only time I've seen this with 1.8.5 is when I've purposely replicated it after reading your post. After sending that email I had a look at the logs of our production server which are running 1.8.3.2 and I can see no occurrence of those errors. Could it be possible that this crept in when the fix for the sip channel locking bug was fixed? -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- 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 login notification
Hello, Is there a way to either store login/logout agent information in a database or at least send an email when an agent logs in or out of a queue? Thanks, Michael -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue agent login notification
you can monitor queue_log file for ADDMEMBER or REMOVEMEMBER events. from that point on, you can store them or take any other action. the other way is to use AMI an monitor for Agent login/logoff events On Aug 12, 2011, at 7:06 AM, Michael wrote: Hello, Is there a way to either store login/logout agent information in a database or at least send an email when an agent logs in or out of a queue? Thanks, Michael -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Interrupting a call in progress?
Is it possible to butt in on a call in progress and play a message to one party, without disconnecting the call? (Anyone with fond [or not-so-fond] memories of the old GPO payphones will remember the pips used to indicate that a coin needed to be inserted to keep the call alive.) Why do I want to do this? My phone company allows calls to landlines at a flat rate for up to one hour, then begins charging per second *including* the hour you've already used. If you hang up and redial anytime up to the 3599th second, you get another hour for cheap; but if you go past the hour, you get penalised. So what I want to do is, play an announcement to alert someone on an internal extension (either SIP for my modern phones or DAHDI for my antique ones) speaking to an external number (DAHDI) after 55 minutes. I *could* just set an absolute timeout in my outgoing context, so calls get cut off just before the 1 hour limit; but this is somewhat inelegant from the user's point of view. I'm thinking I will need to run an AGI script once the call is bridged, to capture the channel name; then schedule another job to start at a predetermined time which will check to see if the channel is still active and, if so, butt in with the announcement. So if there's a method of interjecting on a call from the Unix command line (or from within a Perl script), I could take it the rest of the way from there. -- 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] Problem setting for incoming termination
On 08/11/2011 02:03 AM, Jim Boykin wrote:
We have difficulty setting up the incoming termination for our
clients. Both the ends are using asterisk. The problem is unless we
use fromuser at client end, it does not work properly as expected.
Below is a configuration at our end. The problem is that whenever call
is received from the client, it goes to default context instead of
'dallas' context. Also, the ${CDR(accountcode)} variable remains
empty. Now, If we set fromuser field at the client end, then
everything starts working, however, in that case, it overrides the
callerid.
This is a known and well-understood problem caused by the method that
Asterisk users for SIP authentication; the 'From' header in the incoming
INVITE is used *both* for determining which user is placing the call and
for Caller ID. As you note, if you have the real Caller ID in that
header, then Asterisk can't use it for matching to a user in sip.conf,
and thus can't authenticate the call properly.
The solution for this is to use 'sendrpid' on the sending end and
'trustrpid' on the receiving end; this will configure Asterisk to
transfer the Caller ID information in a Remote-Party-ID (or
P-Asserted-Identity, depending on the version you are using) header,
allowing the From header to be used solely for authentication.
--
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Re: [asterisk-users] Problem setting for incoming termination
12 aug 2011 kl. 14:51 skrev Kevin P. Fleming:
On 08/11/2011 02:03 AM, Jim Boykin wrote:
We have difficulty setting up the incoming termination for our
clients. Both the ends are using asterisk. The problem is unless we
use fromuser at client end, it does not work properly as expected.
Below is a configuration at our end. The problem is that whenever call
is received from the client, it goes to default context instead of
'dallas' context. Also, the ${CDR(accountcode)} variable remains
empty. Now, If we set fromuser field at the client end, then
everything starts working, however, in that case, it overrides the
callerid.
This is a known and well-understood problem caused by the method that
Asterisk users for SIP authentication; the 'From' header in the incoming
INVITE is used *both* for determining which user is placing the call and for
Caller ID. As you note, if you have the real Caller ID in that header, then
Asterisk can't use it for matching to a user in sip.conf, and thus can't
authenticate the call properly.
The solution for this is to use 'sendrpid' on the sending end and 'trustrpid'
on the receiving end; this will configure Asterisk to transfer the Caller ID
information in a Remote-Party-ID (or P-Asserted-Identity, depending on the
version you are using) header, allowing the From header to be used solely for
authentication.
Or stop using type=user and type=friend, and stick to type=peer and ASterisk
will only match on IP+port address.
/O
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[asterisk-users] Queuing outgoing calls
Hi All, Usually we need to queue calls coming from outside our system and for that we use queues.conf, in this case we have a lot of employees that are using POTS (Dahdi or zap channels) and we want to make them go by order since we have limited outgoing lines, does anybody have a clue what to use in this case, is queues.conf will still be useful in the case figure? -- Khalid Touati Network Administrator at Endosoft, LLC CCNA -- _ -- 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] Queuing outgoing calls
You can use call-queueing to accomplish this. When your employee dials the number (555-1212 for demonstration purposes), instead of going directly out, the call goes to /var/spool/asterisk/outgoing as an entry. When this entry comes up, the employee gets a call-back/connect to his/her party. You would need to provide a 911 and/or executive loophole however. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of khalid touati Sent: Friday, August 12, 2011 8:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Queuing outgoing calls Hi All, Usually we need to queue calls coming from outside our system and for that we use queues.conf, in this case we have a lot of employees that are using POTS (Dahdi or zap channels) and we want to make them go by order since we have limited outgoing lines, does anybody have a clue what to use in this case, is queues.conf will still be useful in the case figure? -- Khalid Touati Network Administrator at Endosoft, LLC CCNA -- _ -- 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] Queuing outgoing calls
Hi Danny, Thnks for your response but I googled call-queueing with no success, are your referring to the concept or a third party application or an Asterisk function..., can you please specify? On Fri, Aug 12, 2011 at 10:03 AM, Danny Nicholas da...@debsinc.com wrote: You can use “call-queueing” to accomplish this. When your employee dials the number (555-1212 for demonstration purposes), instead of going directly out, the call goes to /var/spool/asterisk/outgoing as an entry. When this entry comes up, the employee gets a call-back/connect to his/her party. You would need to provide a 911 and/or executive loophole however. ** ** -- _ -- 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 -- Khalid Touati Network Administrator at Endosoft, LLC CCNA -- _ -- 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 files in /var/spool/asterisk/outgoing
Hi !
I have a python script that create and move .call files to
/var/spool/asterisk/outgoing
Sometimes...(in this case after 500 successfull calls) Asterisk don´t make
the calls and the .call files are in the outgoing forever...
Any Ideas?
I'm using Asterisk 1.4.22 (in 1.4.36 was the same behavior)
In my python script I move .call files using ...
import shutil
shutil.move('/var/tmp/1.call','/var/spool/asterisk/outgoing/1.call')
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Re: [asterisk-users] .call files in /var/spool/asterisk/outgoing
On Fri, Aug 12, 2011 at 12:23:22PM -0300, equis software wrote:
shutil.move('/var/tmp/1.call','/var/spool/asterisk/outgoing/1.call')
Are both /var/tmp and /var/spool/asterisk/outgoing on the same
filesystem?
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Re: [asterisk-users] .call files in /var/spool/asterisk/outgoing
Yes, same server, same filesystem...
On Fri, Aug 12, 2011 at 12:26 PM, Roger Burton West ro...@firedrake.orgwrote:
On Fri, Aug 12, 2011 at 12:23:22PM -0300, equis software wrote:
shutil.move('/var/tmp/1.call','/var/spool/asterisk/outgoing/1.call')
Are both /var/tmp and /var/spool/asterisk/outgoing on the same
filesystem?
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Re: [asterisk-users] .call files in /var/spool/asterisk/outgoing
On Fri, Aug 12, 2011 at 12:27:45PM -0300, equis software wrote: Yes, same server, same filesystem... I don't do Python, but a web search for shutil.move suggests that it doesn't reliably use the rename syscall. Might be worth shelling out to your system's mv command. R -- _ -- 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 files in /var/spool/asterisk/outgoing
Also, keep in mind that the spooling mechanism has mechanical limits based on processor speed, line capacity, etc. If I were doing 500 calls, I would use sleep to space the starting of the calls (maybe 5 or 15 second intervals). -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Roger Burton West Sent: Friday, August 12, 2011 10:32 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] .call files in /var/spool/asterisk/outgoing On Fri, Aug 12, 2011 at 12:27:45PM -0300, equis software wrote: Yes, same server, same filesystem... I don't do Python, but a web search for shutil.move suggests that it doesn't reliably use the rename syscall. Might be worth shelling out to your system's mv command. R -- _ -- 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] Conference calls through web-interface with moderation using Asterisk?
Good Morning, I have been researching this for a while, basically I'd like to have a website with the following functionality: • One-click call-in to show (after setting username, best-case scenario: sign-in through Drupal, use that name for conference-call) • Web-interface only (Flash/Flex, Javascript/JQuery or Java), without any additional software/addons/plugins to install • Moderation: host of conference call can quieten/mute or even kick people from the conference call if they're being rowdy So far I have setup an IceCAST server, broadcasting through edcast in an mp3 stream. Viewers of my website can now listen-in on the /radio/ sub-page. How do I setup the aforementioned [3] features using Asterisk? — Do I need [Free, Open-Source] products other than Asterisk to get this done, if so, which? Thanks for all suggestions, Alec Taylor -- _ -- 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 files in /var/spool/asterisk/outgoing
I made 500 calls but not simultaneously. My script checks that there are no more than 3 .call files in the outgoing. I change in my python script, now move file with os.system... import os os.system (mv+ + tmpFile + + callFile) see what happens... On Fri, Aug 12, 2011 at 12:40 PM, Danny Nicholas da...@debsinc.com wrote: Also, keep in mind that the spooling mechanism has mechanical limits based on processor speed, line capacity, etc. If I were doing 500 calls, I would use sleep to space the starting of the calls (maybe 5 or 15 second intervals). -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Roger Burton West Sent: Friday, August 12, 2011 10:32 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] .call files in /var/spool/asterisk/outgoing On Fri, Aug 12, 2011 at 12:27:45PM -0300, equis software wrote: Yes, same server, same filesystem... I don't do Python, but a web search for shutil.move suggests that it doesn't reliably use the rename syscall. Might be worth shelling out to your system's mv command. R -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] .call files in /var/spool/asterisk/outgoing
Another thought - when a call in /V/S/A/O fails, the file gets appended with call info and retry occurs. You might want to write a second Python script to check for and possibly purge failed call files. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of equis software Sent: Friday, August 12, 2011 11:06 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] .call files in /var/spool/asterisk/outgoing I made 500 calls but not simultaneously. My script checks that there are no more than 3 .call files in the outgoing. I change in my python script, now move file with os.system... import os os.system (mv+ + tmpFile + + callFile) see what happens... On Fri, Aug 12, 2011 at 12:40 PM, Danny Nicholas da...@debsinc.com wrote: Also, keep in mind that the spooling mechanism has mechanical limits based on processor speed, line capacity, etc. If I were doing 500 calls, I would use sleep to space the starting of the calls (maybe 5 or 15 second intervals). -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Roger Burton West Sent: Friday, August 12, 2011 10:32 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] .call files in /var/spool/asterisk/outgoing On Fri, Aug 12, 2011 at 12:27:45PM -0300, equis software wrote: Yes, same server, same filesystem... I don't do Python, but a web search for shutil.move suggests that it doesn't reliably use the rename syscall. Might be worth shelling out to your system's mv command. R -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queuing outgoing calls
The .call file can connect an internal number to an outside number
Look at this sample
Channel: DAHDI/R1/5551212
MaxRetries: 2
# Retry in 5 min
RetryTime: 300
WaitTime: 45
Context: outgoing
Extension:100
Priority: 1
This sample call makes a call on DAHDI using Round Robin Group 1. If the
call can be made, it connects to internal extension 100. So instead of your
employee dialing 5551212 directly, they dial 1234 and enter 5551212 as the
number to be dialed. When a line becomes available and the call goes
through, 100 is bridged in and the call is done
Exten = 1234,1,read(numtodial,enternum,10,skip,1,10)
Exten = 1234,2,AGI(makecall.agi,${EXTEN},${numtodial})
Exten = 1234,3,hangup()
Makecall.agi
#!/bin/sh
echo extension: $1 call1.tmp
echo maxtries: 3 call1.tmp
echo retrytime: 300 call1.tmp
echo Channel: DAHDI/R1/$2 call1.tmp
echo Priority: 1 call1.tmp
chmod +x call1.tmp
mv call1.tmp /var/spool/asterisk/outgoing
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of khalid touati
Sent: Friday, August 12, 2011 9:56 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Queuing outgoing calls
Hi Danny,
Thnks for your response but I googled call-queueing with no success, are
your referring to the concept or a third party application or an Asterisk
function..., can you please specify?
On Fri, Aug 12, 2011 at 10:03 AM, Danny Nicholas da...@debsinc.com wrote:
You can use call-queueing to accomplish this. When your employee dials
the number (555-1212 for demonstration purposes), instead of going directly
out, the call goes to /var/spool/asterisk/outgoing as an entry. When this
entry comes up, the employee gets a call-back/connect to his/her party. You
would need to provide a 911 and/or executive loophole however.
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Network Administrator at Endosoft, LLC
CCNA
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Re: [asterisk-users] Queuing outgoing calls
On Fri, 12 Aug 2011, Danny Nicholas wrote:
Exten = 1234,2,AGI(makecall.agi,${EXTEN},${numtodial})
Makecall.agi
#!/bin/sh
echo extension: $1 call1.tmp
echo maxtries: 3 call1.tmp
echo retrytime: 300 call1.tmp
echo Channel: DAHDI/R1/$2 call1.tmp
echo Priority: 1 call1.tmp
From a 'best practices' standpoint I think it would be better to use
system() to execute this script since it is (obviously) not really an AGI.
I'm guessing that system() would be slightly more efficient than agi().
Both require a process creation, but agi() requires (slightly) more
Asterisk resources in setting up the AGI environment.
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Re: [asterisk-users] Queuing outgoing calls
Hey Danny thanks a bunch! I really appreciate that.
Thank you Steve!
On Fri, Aug 12, 2011 at 3:05 PM, Danny Nicholas da...@debsinc.com wrote:
The .call file can connect an internal number to an outside number
Look at this sample
Channel: DAHDI/R1/5551212
MaxRetries: 2
# Retry in 5 min
RetryTime: 300
WaitTime: 45
Context: outgoing
Extension:100
Priority: 1
** **
This sample call makes a call on DAHDI using Round Robin Group 1. If the
call can be made, it connects to internal extension 100. So instead of your
employee dialing 5551212 directly, they dial 1234 and enter 5551212 as the
number to be dialed. When a line becomes available and the call goes
through, 100 is bridged in and the call is done
** **
Exten = 1234,1,read(numtodial,enternum,10,skip,1,10)
Exten = 1234,2,AGI(makecall.agi,${EXTEN},${numtodial})
Exten = 1234,3,hangup()
** **
Makecall.agi
#!/bin/sh
echo extension: $1 call1.tmp
echo maxtries: 3 call1.tmp
echo retrytime: 300 call1.tmp
echo Channel: DAHDI/R1/$2 call1.tmp
echo Priority: 1 call1.tmp
chmod +x call1.tmp
mv call1.tmp /var/spool/asterisk/outgoing
** **
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *khalid touati
*Sent:* Friday, August 12, 2011 9:56 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Queuing outgoing calls
** **
Hi Danny,
Thnks for your response but I googled call-queueing with no success, are
your referring to the concept or a third party application or an Asterisk
function..., can you please specify?
On Fri, Aug 12, 2011 at 10:03 AM, Danny Nicholas da...@debsinc.com
wrote:
You can use “call-queueing” to accomplish this. When your employee dials
the number (555-1212 for demonstration purposes), instead of going directly
out, the call goes to /var/spool/asterisk/outgoing as an entry. When this
entry comes up, the employee gets a call-back/connect to his/her party. You
would need to provide a 911 and/or executive loophole however.
** **
** **
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Khalid Touati
Network Administrator at Endosoft, LLC
CCNA
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Khalid Touati
Network Administrator at Endosoft, LLC
CCNA
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Re: [asterisk-users] .call files in /var/spool/asterisk/outgoing
Hello, Check if file is owned by asterisk user. Also, don't directly create in to /var/spool/asterisk/outgoing/ Create in somewhere else first and then move file to outgoing folder. Good luck. On Fri, Aug 12, 2011 at 7:09 PM, Danny Nicholas da...@debsinc.com wrote: Another thought – when a call in /V/S/A/O fails, the file gets appended with call info and retry occurs. You might want to write a second Python script to check for and possibly purge failed call files. ** ** *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *equis software *Sent:* Friday, August 12, 2011 11:06 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] .call files in /var/spool/asterisk/outgoing ** ** I made 500 calls but not simultaneously. My script checks that there are no more than 3 .call files in the outgoing. I change in my python script, now move file with os.system... import os os.system (mv+ + tmpFile + + callFile) see what happens... On Fri, Aug 12, 2011 at 12:40 PM, Danny Nicholas da...@debsinc.com wrote: Also, keep in mind that the spooling mechanism has mechanical limits based on processor speed, line capacity, etc. If I were doing 500 calls, I would use sleep to space the starting of the calls (maybe 5 or 15 second intervals). -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Roger Burton West Sent: Friday, August 12, 2011 10:32 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] .call files in /var/spool/asterisk/outgoing* *** On Fri, Aug 12, 2011 at 12:27:45PM -0300, equis software wrote: Yes, same server, same filesystem... I don't do Python, but a web search for shutil.move suggests that it doesn't reliably use the rename syscall. Might be worth shelling out to your system's mv command. R -- _ -- 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 -- _ -- 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] Message prints even if verbose level is Zero
In 1.8, somebody left a message that shows up like this Remotely bridging SIP/Client.XX.XX.XX.125-00010456 and SIP/XX.XXX.XX.X-00010457 It could be also Local Bridging The point is that this message should not print in the console unless the verbose level reaches some level. Never at level zero. It should be a notice, etc. When there is a lot of traffic, this message consumes CPU unnecessarily. -- _ -- 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] One way audio when using originate...
We are having a problem when trying to use originate or AMI to make a call. We have an Asterisk 1.8.5.0 server which uses a SIP provider to call the PSTN. When dialing from IP phones everything works fine. When you try making the call with originate, AMI or a call file then the remote person can hear you but you cannot hear them. Why would it behave differently when dialing from a phone? The server is behind NAT and uses externaddr to set the external IP (static). Anyone had any experience with this? Here is my (edited) sip.conf entry: [libre-8793] defaultuser=123456789 secret=X fromuser=123456789 trustrpid=yes sendrpid=yes type=peer fromdomain=i2next.com.mx host=i2next.com.mx nat=yes qualify=no insecure=port,invite directmedia=no disallow=all allow=g729 -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Message prints even if verbose level is Zero
On 08/12/2011 03:23 PM, CDR wrote: In 1.8, somebody left a message that shows up like this Remotely bridging SIP/Client.XX.XX.XX.125-00010456 and SIP/XX.XXX.XX.X-00010457 It could be also Local Bridging The point is that this message should not print in the console unless the verbose level reaches some level. Never at level zero. It should be a notice, etc. When there is a lot of traffic, this message consumes CPU unnecessarily. Posting this here isn't likely to result in the code getting changed, unless a developer just happens to see it. Please open an issue in the issue tracker so it won't be forgotten. Thanks. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
