[asterisk-users] Dialing multiple endpoints and CallerID presentation
Hi, I've got the following use case where I want to simultaneously dial 2 endpoints that both need different CallerID presentation. How can I do that, from the dialplan preferably ? For instance, let say phone A needs to both dial B, an internal SIP phone and C, a cell phone reachable through a DAHDI span from a an Asterisk system where : 1. users can use 4-digits short numbers to reach other internal phones. 2. calls going out through the DAHDI span, must have CallerIDs presented without any prefix. Ideally, CallerID should be presented : 1- with 4-digits for internal phones 2- with 10-digits for external phones so that both phones can return the call without re-dialing. Suggestions ? A is 1234 alias DID 051234 B is 5678 C is 0123456789 I was thinking of using something like this: Dial(SIP/5678option_to_present_1234_to_calleeDAHDI/g1option_to_present_051234/0123456789) What could be option_to_present_1234_to_callee and option_to_present_051234 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] Create ou update values ins ASTB when Asterisk is stopped
Hi, The title says it all : is it possible to create ou update values ins ASTB when Asterisk is stopped ? When Asterisk is running, I'm using : asterisk -rx database put Foo Bar 1 Can I do that when Asterisk is stopped, using some program that can read and write in ASTDB (I'm using Debian Squeeze). Will this change when ASTDB will be moving from Berkeley DB to SQLite 3 ? 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
Re: [asterisk-users] Dialing multiple endpoints and CallerID presentation
Alternative work around to this could be: 1- Make two different dialplan extensions. One to dial DAHDI numbers with setting for DAHDI and other extension for SIP dialing. Both extensions setting different CallerID presentation 2- Create a queue with Local extensions as static members (strategy=ringall) So whenever you want to dial to both B, and C location use the queue dial-out. I think it should work. On Mon, Aug 29, 2011 at 12:15 PM, Olivier oza_4...@yahoo.fr wrote: Hi, I've got the following use case where I want to simultaneously dial 2 endpoints that both need different CallerID presentation. How can I do that, from the dialplan preferably ? For instance, let say phone A needs to both dial B, an internal SIP phone and C, a cell phone reachable through a DAHDI span from a an Asterisk system where : 1. users can use 4-digits short numbers to reach other internal phones. 2. calls going out through the DAHDI span, must have CallerIDs presented without any prefix. Ideally, CallerID should be presented : 1- with 4-digits for internal phones 2- with 10-digits for external phones so that both phones can return the call without re-dialing. Suggestions ? A is 1234 alias DID 051234 B is 5678 C is 0123456789 I was thinking of using something like this: Dial(SIP/5678option_to_present_1234_to_calleeDAHDI/g1option_to_present_051234/0123456789) What could be option_to_present_1234_to_callee and option_to_present_051234 Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Presence for channels other than SIP.
Hi How to get the presence status of channels than SIP like Phone,Dahdi ,gsm and etc. I have checked the DEVICE_STATE function in dialplan but it shows only SIP channels status may be IAX too ,for other type channels(Phone,Dahdi,gsm) it is not showing anything.,And I tried hint too then also same result. Is this feature available in asterisk ? Thanks Nikhil -- _ -- 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] Dialing multiple endpoints and CallerID presentation
Hi,
Use the local channel
Dial(Local/@contextinternallocal/b@contextexternal)
In the internal context you set CALLERID(num) to the internal extension
and then dial the SIP
exten = ,1,Set(CALLERDI(num)=${EXTEN})
same = n,Dial(SIP/${EXTEN})
In the external context do almost the same but dial DAHDI
exten = bb,1,Set(CALLERDI(num)=051234)
same = n,Dial(DAHDI/g1/0123456789)
Regards,
Michel.
Op 29-08-11 09:15, Olivier schreef:
Hi,
I've got the following use case where I want to simultaneously dial 2
endpoints that both need different CallerID presentation.
How can I do that, from the dialplan preferably ?
For instance, let say phone A needs to both dial B, an internal SIP
phone and C, a cell phone reachable through a DAHDI span from a an
Asterisk system where :
1. users can use 4-digits short numbers to reach other internal phones.
2. calls going out through the DAHDI span, must have CallerIDs
presented without any prefix.
Ideally, CallerID should be presented :
1- with 4-digits for internal phones
2- with 10-digits for external phones
so that both phones can return the call without re-dialing.
Suggestions ?
A is 1234 alias DID 051234
B is 5678
C is 0123456789
I was thinking of using something like this:
Dial(SIP/5678option_to_present_1234_to_calleeDAHDI/g1option_to_present_051234/0123456789)
What could be option_to_present_1234_to_callee and
option_to_present_051234
Regards
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk is delaying DTMF (SIP INFO) relay in MeetMe
Hello List, We are using 'F' parameter in meetme Dialplan application to broadcast SIP INFO (1 and 0) as DTMF tone to all the participants. The DTMF configuration for all the connected SIP clients is SIP INFO. The problem we are seeing, asterisk is taking some time to broadcast the SIP INFO message to all the participants from the time of its appearance. The time latency varies from 1.5 sec to 6 sec. We have activated the highest debug and verbose level but we are not able to track down the problem. Please help us out to overcome this problem as 6 sec latency is not acceptable in real-time scenarios. Also if possible let us know (technically), whether it is a know issue in asterisk. Regards, Rajib Siemens Ltd. -- _ -- 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] seeding an originated number in a SIP phone [was: Re: Thunderbird extension using AMI to dial]
On 08/28/2011 01:56 AM, Tzafrir Cohen wrote: On Thu, Aug 25, 2011 at 07:36:53PM +0100, Chris Hastie wrote: Hi I've just added direct support for AMI to a forthcoming version of TBDialOut, a Thunderbird extension for dialling direct from Thunderbird's address book. If anyone fancies testing it I'd be grateful for any feedback. If you feel like casting a critical eye over the code, or doing some translating, even better. AMI support is available in TBDialOut 1.7.0pre1, which can be found either at http://www.oak-wood.co.uk/tbdialout/ or from the 'Development channel' at the bottom of the page at https://addons.mozilla.org/en-US/thunderbird/addon/tbdialout/ We already have a dialer script (sent to this list a while ago) so it's good to see that this extension support that simpler option as well (I don't use ThunderBird, as you can see. Some others in the office do use it). One followup question: I originate a call from a SIP phone to some remote number. The problem is that the number will not show up properly in the list of outgoing calls for the phone. Any idea how to fix this (for whatever SIP phone)? You aren't originating a call *from* the phone (that would require some sort of API into the phone itself to make it place a call). You are originating a call *to* the phone and also to another endpoint; as far as the SIP phone is concerned, this is an incoming call. I've never seen discussion of a desire to provide a method for an incoming call to be treated as if the endpoint had placed the call itself in any of the SIP discussion lists I frequent... so I'm pretty sure there's no standard way to do this. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Possible Bug? .call files executing multiple times
On 08/19/2011 09:14 AM, Brandon Phelps wrote: Hello all, We are setting up an auto-dialer to call customers based on the opening of tickets in our internal ticketing system. Everything is going fine so far except for one snag: To test the system we are implementing I am manually moving .call files into the /var/spool/asterisk/outgoing directory like this: asterisk@dialerdev:~# cp test5703.call /tmp/test.call mv /tmp/test.call /var/spool/asterisk/outgoing/ This works great and the call is immediately started, however more often than not (ie. not all the time, but most of the time) after answering the call or rejecting it (sending it to voicemail), another call is performed using the same file. I notice that when a call is initiated the .call file is not removed immediately. Instead, asterisk waits until the call is completed before removing the call file, so it seems like 5-10 seconds into the call since the .call file still exists another call is placed. Any advice on how we can avoid this situation and ensure that only one call is made per .call file? The OS is Ubuntu 11.04 server and we're running Asterisk 1.8. Thanks, Sorry to bring this back up but I am still having this issue and haven't had any luck resolving it. It should be noted that the .call files in question are set to MaxRetries: 0, and simply connect the call to the 's' extension in a custom context. From there the context is pretty complicated, running some AGI scripts along with some dealing with user input, basically a simple IVR. Any help would be appreciated. Thanks, Brandon -- _ -- 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] Looking for ideas for nice **Asterisk** home phone system
On Sat, 2011-08-27 at 09:31 +0100, Alan Lord (News) wrote: On 26/08/11 12:28, linux guy wrote: Great discussion, all of it. Thanks, people. How much power does the home asterisk box need ? Not much :-) I've been running our phone system and home media/storage network on a VIA C7 cpu based home build that I *downclocked* to 1Ghz from 1.2Ghz for about three years now. Al I've been running the house phone system on a re-purposed Seagate Dockstar with a 4G USB stick for over a year now. FreePBX, hylafax, iaxmodem, and 1.4 on Ubuntu. Wish I could still buy these buggers - I got this one for $30! j -- _ -- 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] Possible Bug? .call files executing multiple times
Can you post the .call file (with called number blacked out) before call and after 1-2 calls? (file 1 should be before you mv to /v/s/a/o, file 2 should be from /v/s/a/o). -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Brandon Phelps Sent: Monday, August 29, 2011 8:45 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Possible Bug? .call files executing multiple times On 08/19/2011 09:14 AM, Brandon Phelps wrote: Hello all, We are setting up an auto-dialer to call customers based on the opening of tickets in our internal ticketing system. Everything is going fine so far except for one snag: To test the system we are implementing I am manually moving .call files into the /var/spool/asterisk/outgoing directory like this: asterisk@dialerdev:~# cp test5703.call /tmp/test.call mv /tmp/test.call /var/spool/asterisk/outgoing/ This works great and the call is immediately started, however more often than not (ie. not all the time, but most of the time) after answering the call or rejecting it (sending it to voicemail), another call is performed using the same file. I notice that when a call is initiated the .call file is not removed immediately. Instead, asterisk waits until the call is completed before removing the call file, so it seems like 5-10 seconds into the call since the .call file still exists another call is placed. Any advice on how we can avoid this situation and ensure that only one call is made per .call file? The OS is Ubuntu 11.04 server and we're running Asterisk 1.8. Thanks, Sorry to bring this back up but I am still having this issue and haven't had any luck resolving it. It should be noted that the .call files in question are set to MaxRetries: 0, and simply connect the call to the 's' extension in a custom context. From there the context is pretty complicated, running some AGI scripts along with some dealing with user input, basically a simple IVR. Any help would be appreciated. Thanks, Brandon -- _ -- 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] Scheduled Maintenance for Asterisk Project community services
On Thursday, September 1st, 2011, the Asterisk community services listed below will be undergoing maintenance (power distribution upgrades in the cabinet where the servers are located). The services will be shut down at approximately 11:30 PM CDT, and will return no later than 6:00 AM CDT on September 2nd. We apologize in advance for any inconvenience this may cause. The affected services are: downloads.digium.com downloads.asterisk.org bamboo.asterisk.org git.asterisk.org code.asterisk.org packages.asterisk.org svn.asterisk.org issues.asterisk.org reviewboard.asterisk.org svnview.asterisk.org wiki.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
[asterisk-users] Dragging the dialup customers along, possible?
We have an Ascend Max router that has a PRI plugged into it, providing our current dialup users with web access. The PRI is no longer cost effective, so I've been tasked with converting it to something cheaper. We added a DID to our (existing) asterisk system (we have a couple dozen voip customers). We added an Adtran 908 to convert the VoIP signal into a virtual PRI for our MAX router to handle dialup calls. When dialing into the number with a modem, MAX sees the call, picks up and apparently tries to negotiate it, but eventually disconnects. It HAS, however, twice, successfully connected the call for a short time, but no browsing was possible. I've done some debugging output on the Adtran which seems to indicate that an RTP BYE command is received: TM.T01 01 SipTM_Connected rcvd SIP call-leg request: BYE ... This is the first difference between a debug output where the call connected and one that does not work. This is the one that doesn't work. TL;DR - Is it possible to do dial-up through Asterisk? Or is it like t38 faxing that is 'possible' to get working, but not if you have other job responsibilities? Thanks! -- Aaron Krohn Web Force Systems Business Office: 131 Dillmont Drive, Suite 201 Columbus, OH 43235 Direct: 614-384-0019Fax: 614-785-0871 Tech Support / Help Desk Direct: 614-384-0020 -- _ -- 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] http manager getconfig crashes on reading large files/categories (50+ lines)
When the following HTTP request is processed and the result contains more than 50 lines asterisk will crash every time. The file being read is irrelevant. http://localhost/ajam/rawman?action=getconfigfilename=sip.conf Has anyone experienced this problem. I did file a bug report https://issues.asterisk.org/jira/browse/ASTERISK-18361 -- Marvin Horst -- _ -- 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] Dragging the dialup customers along, possible?
From what you are asking it appears that you are trying to run similar to a fax (modulation and demodulation) over VoIP. Try again, the fact that you succeeded twice was pure luck, and as far as I understand that didn't even work out. Switch back to TDM. Your dial up modems want that magic thing called timing and no jitter that only TDM will give you. On Mon, Aug 29, 2011 at 2:56 PM, Aaron Krohn akr...@ewebforce.net wrote: We have an Ascend Max router that has a PRI plugged into it, providing our current dialup users with web access. The PRI is no longer cost effective, so I've been tasked with converting it to something cheaper. We added a DID to our (existing) asterisk system (we have a couple dozen voip customers). We added an Adtran 908 to convert the VoIP signal into a virtual PRI for our MAX router to handle dialup calls. When dialing into the number with a modem, MAX sees the call, picks up and apparently tries to negotiate it, but eventually disconnects. It HAS, however, twice, successfully connected the call for a short time, but no browsing was possible. I've done some debugging output on the Adtran which seems to indicate that an RTP BYE command is received: TM.T01 01 SipTM_Connected rcvd SIP call-leg request: BYE ... This is the first difference between a debug output where the call connected and one that does not work. This is the one that doesn't work. TL;DR - Is it possible to do dial-up through Asterisk? Or is it like t38 faxing that is 'possible' to get working, but not if you have other job responsibilities? Thanks! -- Aaron Krohn Web Force Systems Business Office: 131 Dillmont Drive, Suite 201 Columbus, OH 43235 Direct: 614-384-0019 Fax: 614-785-0871 Tech Support / Help Desk Direct: 614-384-0020 -- _ -- 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] Dragging the dialup customers along, possible?
It is possible to use Asterisk as a dialup PPP server, but only if you are doing PRI between the telco and Asterisk (see core show application DAHDIRAS). You could bring analog POTS lines into a dialup server (Portmaster maybe?) if PRI is too expensive. Can outsource your dialup customers to a national network? You could simply stop providing dialup service. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of C F Sent: Monday, August 29, 2011 4:18 PM To: t...@ewebforce.net; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Dragging the dialup customers along, possible? From what you are asking it appears that you are trying to run similar to a fax (modulation and demodulation) over VoIP. Try again, the fact that you succeeded twice was pure luck, and as far as I understand that didn't even work out. Switch back to TDM. Your dial up modems want that magic thing called timing and no jitter that only TDM will give you. On Mon, Aug 29, 2011 at 2:56 PM, Aaron Krohn akr...@ewebforce.net wrote: We have an Ascend Max router that has a PRI plugged into it, providing our current dialup users with web access. The PRI is no longer cost effective, so I've been tasked with converting it to something cheaper. We added a DID to our (existing) asterisk system (we have a couple dozen voip customers). We added an Adtran 908 to convert the VoIP signal into a virtual PRI for our MAX router to handle dialup calls. When dialing into the number with a modem, MAX sees the call, picks up and apparently tries to negotiate it, but eventually disconnects. It HAS, however, twice, successfully connected the call for a short time, but no browsing was possible. I've done some debugging output on the Adtran which seems to indicate that an RTP BYE command is received: TM.T01 01 SipTM_Connected rcvd SIP call-leg request: BYE ... This is the first difference between a debug output where the call connected and one that does not work. This is the one that doesn't work. TL;DR - Is it possible to do dial-up through Asterisk? Or is it like t38 faxing that is 'possible' to get working, but not if you have other job responsibilities? -- _ -- 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] Dragging the dialup customers along, possible?
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of C F Sent: Monday, August 29, 2011 3:18 PM To: t...@ewebforce.net; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Dragging the dialup customers along, possible? From what you are asking it appears that you are trying to run similar to a fax (modulation and demodulation) over VoIP. Try again, the fact that you succeeded twice was pure luck, and as far as I understand that didn't even work out. Switch back to TDM. Your dial up modems want that magic thing called timing and no jitter that only TDM will give you. === This is more of a whimsical statement than a scientific one, but I would think in today's world, there would be a real small box that would take in IP and put out TDM with good timing with a moderate buffering window. Obviously, the IP has to actually get to the box in a timely fashion, like today , but a TDM circuit has to be up also. A box that would take in IP data..., look for valid ascii, and otherwise put out TDM modem tones with no data content for 1 second and then pick up the data as it catches up. Better a laggy modem connection than no data at all. CF -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Possible Bug? .call files executing multiple times
Here is the contents of the .call file. The file is the same before the move as after (I did a cat on the file after the move, while the phone was ringing a second time): Channel: Local/5703@ext-main Callerid: MyCompany 8005551234 Set: TicketNumber=100 Set: CallerID_Num=8005551234 Set: CALLSTATUS=0 Context: ext-autodialer MaxRetries: 0 WaitTime: 45 Extension: s Priority: 1 We have tried using a SIP channel as well (as opposed to Local) with the same results. The s extension of ext-autodialer runs an AGI script which makes use of those Set: variables. I can most easily reproduce the problem by simply not answering the call. After 2 or 3 rings line 2 on the phone lights up indicating another call. If I reject the first call and answer the second call, it's the same script. Also during my most recent test the following happened: 1. I moved file to /var/spool/asterisk/outgoing 2. Phone rang on line 1 3. I let phone continue to ring 4. After 3 rings, line 2 started ringing (another call from the same .call file) 5. I rejected both calls, sending both to voicemail. 6. 6 or 7 seconds after rejecting both calls, the phone rang a 3rd time. 7. I let the phone ring until it was automatically moved to voicemail and finally the .call file was removed. On 08/29/2011 11:00 AM, Danny Nicholas wrote: Can you post the .call file (with called number blacked out) before call and after 1-2 calls? (file 1 should be before you mv to /v/s/a/o, file 2 should be from /v/s/a/o). -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Brandon Phelps Sent: Monday, August 29, 2011 8:45 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Possible Bug? .call files executing multiple times On 08/19/2011 09:14 AM, Brandon Phelps wrote: Hello all, We are setting up an auto-dialer to call customers based on the opening of tickets in our internal ticketing system. Everything is going fine so far except for one snag: To test the system we are implementing I am manually moving .call files into the /var/spool/asterisk/outgoing directory like this: asterisk@dialerdev:~# cp test5703.call /tmp/test.call mv /tmp/test.call /var/spool/asterisk/outgoing/ This works great and the call is immediately started, however more often than not (ie. not all the time, but most of the time) after answering the call or rejecting it (sending it to voicemail), another call is performed using the same file. I notice that when a call is initiated the .call file is not removed immediately. Instead, asterisk waits until the call is completed before removing the call file, so it seems like 5-10 seconds into the call since the .call file still exists another call is placed. Any advice on how we can avoid this situation and ensure that only one call is made per .call file? The OS is Ubuntu 11.04 server and we're running Asterisk 1.8. Thanks, Sorry to bring this back up but I am still having this issue and haven't had any luck resolving it. It should be noted that the .call files in question are set to MaxRetries: 0, and simply connect the call to the 's' extension in a custom context. From there the context is pretty complicated, running some AGI scripts along with some dealing with user input, basically a simple IVR. Any help would be appreciated. Thanks, Brandon -- _ -- 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] Possible Bug? .call files executing multiple times
Also I should note that we use the 'noatime' attribute on the /var filesystem, would this cause the problem below? On 08/29/2011 05:22 PM, Brandon Phelps wrote: Here is the contents of the .call file. The file is the same before the move as after (I did a cat on the file after the move, while the phone was ringing a second time): Channel: Local/5703@ext-main Callerid: MyCompany 8005551234 Set: TicketNumber=100 Set: CallerID_Num=8005551234 Set: CALLSTATUS=0 Context: ext-autodialer MaxRetries: 0 WaitTime: 45 Extension: s Priority: 1 We have tried using a SIP channel as well (as opposed to Local) with the same results. The s extension of ext-autodialer runs an AGI script which makes use of those Set: variables. I can most easily reproduce the problem by simply not answering the call. After 2 or 3 rings line 2 on the phone lights up indicating another call. If I reject the first call and answer the second call, it's the same script. Also during my most recent test the following happened: 1. I moved file to /var/spool/asterisk/outgoing 2. Phone rang on line 1 3. I let phone continue to ring 4. After 3 rings, line 2 started ringing (another call from the same .call file) 5. I rejected both calls, sending both to voicemail. 6. 6 or 7 seconds after rejecting both calls, the phone rang a 3rd time. 7. I let the phone ring until it was automatically moved to voicemail and finally the .call file was removed. On 08/29/2011 11:00 AM, Danny Nicholas wrote: Can you post the .call file (with called number blacked out) before call and after 1-2 calls? (file 1 should be before you mv to /v/s/a/o, file 2 should be from /v/s/a/o). -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Brandon Phelps Sent: Monday, August 29, 2011 8:45 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Possible Bug? .call files executing multiple times On 08/19/2011 09:14 AM, Brandon Phelps wrote: Hello all, We are setting up an auto-dialer to call customers based on the opening of tickets in our internal ticketing system. Everything is going fine so far except for one snag: To test the system we are implementing I am manually moving .call files into the /var/spool/asterisk/outgoing directory like this: asterisk@dialerdev:~# cp test5703.call /tmp/test.call mv /tmp/test.call /var/spool/asterisk/outgoing/ This works great and the call is immediately started, however more often than not (ie. not all the time, but most of the time) after answering the call or rejecting it (sending it to voicemail), another call is performed using the same file. I notice that when a call is initiated the .call file is not removed immediately. Instead, asterisk waits until the call is completed before removing the call file, so it seems like 5-10 seconds into the call since the .call file still exists another call is placed. Any advice on how we can avoid this situation and ensure that only one call is made per .call file? The OS is Ubuntu 11.04 server and we're running Asterisk 1.8. Thanks, Sorry to bring this back up but I am still having this issue and haven't had any luck resolving it. It should be noted that the .call files in question are set to MaxRetries: 0, and simply connect the call to the 's' extension in a custom context. From there the context is pretty complicated, running some AGI scripts along with some dealing with user input, basically a simple IVR. Any help would be appreciated. Thanks, Brandon -- _ -- 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:
Re: [asterisk-users] Possible Bug? .call files executing multiple times
Asterisk has to be able to execute and rewrite the file - the call file is updated in place and when the call is considered successful, removed. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Brandon Phelps Sent: Monday, August 29, 2011 4:28 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Possible Bug? .call files executing multiple times Also I should note that we use the 'noatime' attribute on the /var filesystem, would this cause the problem below? On 08/29/2011 05:22 PM, Brandon Phelps wrote: Here is the contents of the .call file. The file is the same before the move as after (I did a cat on the file after the move, while the phone was ringing a second time): Channel: Local/5703@ext-main Callerid: MyCompany 8005551234 Set: TicketNumber=100 Set: CallerID_Num=8005551234 Set: CALLSTATUS=0 Context: ext-autodialer MaxRetries: 0 WaitTime: 45 Extension: s Priority: 1 We have tried using a SIP channel as well (as opposed to Local) with the same results. The s extension of ext-autodialer runs an AGI script which makes use of those Set: variables. I can most easily reproduce the problem by simply not answering the call. After 2 or 3 rings line 2 on the phone lights up indicating another call. If I reject the first call and answer the second call, it's the same script. Also during my most recent test the following happened: 1. I moved file to /var/spool/asterisk/outgoing 2. Phone rang on line 1 3. I let phone continue to ring 4. After 3 rings, line 2 started ringing (another call from the same .call file) 5. I rejected both calls, sending both to voicemail. 6. 6 or 7 seconds after rejecting both calls, the phone rang a 3rd time. 7. I let the phone ring until it was automatically moved to voicemail and finally the .call file was removed. On 08/29/2011 11:00 AM, Danny Nicholas wrote: Can you post the .call file (with called number blacked out) before call and after 1-2 calls? (file 1 should be before you mv to /v/s/a/o, file 2 should be from /v/s/a/o). -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Brandon Phelps Sent: Monday, August 29, 2011 8:45 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Possible Bug? .call files executing multiple times On 08/19/2011 09:14 AM, Brandon Phelps wrote: Hello all, We are setting up an auto-dialer to call customers based on the opening of tickets in our internal ticketing system. Everything is going fine so far except for one snag: To test the system we are implementing I am manually moving .call files into the /var/spool/asterisk/outgoing directory like this: asterisk@dialerdev:~# cp test5703.call /tmp/test.call mv /tmp/test.call /var/spool/asterisk/outgoing/ This works great and the call is immediately started, however more often than not (ie. not all the time, but most of the time) after answering the call or rejecting it (sending it to voicemail), another call is performed using the same file. I notice that when a call is initiated the .call file is not removed immediately. Instead, asterisk waits until the call is completed before removing the call file, so it seems like 5-10 seconds into the call since the .call file still exists another call is placed. Any advice on how we can avoid this situation and ensure that only one call is made per .call file? The OS is Ubuntu 11.04 server and we're running Asterisk 1.8. Thanks, Sorry to bring this back up but I am still having this issue and haven't had any luck resolving it. It should be noted that the .call files in question are set to MaxRetries: 0, and simply connect the call to the 's' extension in a custom context. From there the context is pretty complicated, running some AGI scripts along with some dealing with user input, basically a simple IVR. Any help would be appreciated. Thanks, Brandon -- _ -- 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
Re: [asterisk-users] USB or Ethernet based FXO device ?
On Sat, 27 Aug 2011 09:31:12 -0600, linux guy linuxguy...@gmail.com wrote: I'm looking for an FXO device to connect to a POTS line that communicates via USB or Ethernet. For USB, AFAIK, there's only the one from Sangoma. All others are Ethernet-based. www.voip-info.org/wiki/view/VoIP+Gateways -- _ -- 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] Dragging the dialup customers along, possible?
On Mon, Aug 29, 2011 at 4:32 PM, Cary Fitch ca...@usawide.net wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of C F Sent: Monday, August 29, 2011 3:18 PM To: t...@ewebforce.net; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Dragging the dialup customers along, possible? From what you are asking it appears that you are trying to run similar to a fax (modulation and demodulation) over VoIP. Try again, the fact that you succeeded twice was pure luck, and as far as I understand that didn't even work out. Switch back to TDM. Your dial up modems want that magic thing called timing and no jitter that only TDM will give you. === This is more of a whimsical statement than a scientific one, but I would think in today's world, there would be a real small box that would take in IP and put out TDM with good timing with a moderate buffering window. Obviously, the IP has to actually get to the box in a timely fashion, like today , but a TDM circuit has to be up also. A box that would take in IP data..., look for valid ascii, and otherwise put out TDM modem tones with no data content for 1 second and then pick up the data as it catches up. So you want to develop the equivalent of T.38 for dial up? Better a laggy modem connection than no data at all. CF -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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
