Re: [asterisk-users] pstn hangs up: MWI no message waiting ??
On Friday 16 January 2009 20:27:57 sean darcy wrote: Tilghman Lesher wrote: On Friday 16 January 2009 17:43:21 sean darcy wrote: Danny Nicholas wrote: Why not do a zap restart instead of restarting asterisk? You could write an AGI to do the ZR when the condition occurred and lines where empty. Yes, a cron job to restart zaptel would cut off any call then existing. But how would I test for it? I can imagine: exten=s,n,ExecIf(some damn thing, System(service dahdi restart)) It's the some damn thing I can't imagine. How do you test if dahdi is acting up? Not a service restart, but a dahdi restart. You can't restart the dahdi service without first stopping Asterisk, anyway. if [ `../asterisk-trunk/contrib/scripts/astcli core show channels | wc -l` = 3 ]; then asterisk -rx 'dahdi restart'; fi Wow. I'll try that tomorrow. Put it as the cmd right after answer(), right? Or maybe, h,1 ? Well anyway, at least I'll be able to receive calls over pstn with dahdi. No, I'd actually recommend that as a cron job. It's basically, restart if idle. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Portech MV-378 with Asterisk
Yes. That's the correct way to do it. Placing # as a rule in callnum forces the Portech to use the number defined in the SIP INVITE packet. Bye. Marco. Marco Signorini INGEGNI Tech S.r.l. http://www.ingegnitech.com http://www.ingegnitechcom/ Pascal Bruno wrote: Sorry for bothering you, but I got it, I just had to put # in callnum! On Sat, Jan 17, 2009 at 1:44 AM, Pascal Bruno tipas...@gmail.com mailto:tipas...@gmail.com wrote: I want to dial out using the sim card. What I did, I have used the SIP channel ex: Channel: SIP/thenum...@mv378 It shows the called is being made in the dialplan, but the number I have entered does not dial, it just goes straight to the specified dialplan extensions. Then what I did, in the Lan to Mobile Table, I put * in url and the number I wanted to dial in call num, then the call was made to that number using the sim card properly. I was wondering if I cannot supply the number to be dialed using an asterisk call file, or do I have to put that number in the Lan to Mobile table. Any help would be appreciated. Thanks On Sat, Jan 17, 2009 at 12:39 AM, Pascal Bruno tipas...@gmail.com mailto:tipas...@gmail.com wrote: Marco, The configs work fine for me. I can receive calls with no problem. Now, were you able to dial using the sim card? I cant figure out how I can do it since asterisk doesnt have a channel to place call through the portech gateway. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] UpdateConfig : Appending line fails
On Friday 16 January 2009 22:43:51 Jose P. Espinal wrote: About UpdateConfig syntax, how did you find out the correct way of sending various sets of parameters? I was looking in google, the ATFOT v2 Book, and nothing showed up. I wrote a patch for a problem with that function last month, and therefore I have become intimately familiar with the code (even though that code wasn't at fault, I had to analyze it to make that determination). -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to transfer a call from one AsteriskServerto another
Are you sure that the TRANSFER is supported by the other side at all? see http://thread.gmane.org/gmane.comp.telephony.pbx.asterisk.devel/15267 Thanks l. 2009/1/16 Paul bulkm...@monafamily.com Yes, this is the first method I tried. The transfer only works if it is done before a media path is set up to the first box (not answered by the IVR). If it is answered then transferred, I get a 500 internal server error back from the ITSP and the call dies. I never see anything hit the second box. -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Lenz Emilitri *Sent:* Friday, January 16, 2009 10:09 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] How to transfer a call from one AsteriskServerto another I guess you already tried this? http://www.voip-info.org/wiki-Asterisk+cmd+Transfer Thanks l. 2009/1/16 Paul bulkm...@monafamily.com I do have it functioning with Dial(). I was looking for a way to completely move the call from the first box though. When using Dial() media moves, but the call is still tied to the first box. In looking at captures when the call is ended, the first box invites out to the ITSP again, then after receiving a 200ok sends a bye. Also while testing, once the call was up on the second box, I stopped Asterisk on the first box which kills the call. -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Lenz Emilitri *Sent:* Friday, January 16, 2009 12:17 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] How to transfer a call from one AsteriskServer to another Why don't you simply Dial() the call to a separate box keeping Asterisk out of the audio path? l. 2009/1/16 Paul bulkm...@monafamily.com Can anyone tell me how I can completely move an established call off of one Asterisk server to another? In our case we have a server with our IVR. Depending upon digits entered, the call can be transferred to any of our other servers depending where the extension or queue reside. We would like to completely move the call off of the first box so we don't tie up resources on it. In our lab we are testing with 1.4.22.1 Our provider which delivers inbound calls to us uses a Sonus gateway. So far, testing has shown that if we transfer the inbound call prior to any media playback, it works. But, if the IVR plays media, then it is failing, with a 500 internal server error being returned. Thanks for any help ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Loway - home of QueueMetrics - http://queuemetrics.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Loway - home of QueueMetrics - http://queuemetrics.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Loway - home of QueueMetrics - http://queuemetrics.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR Rewrite -- Questions to the users
Grey Man greymanv...@gmail.com writes: The trick with transfers is to forget about the src field for billing purposes and make sure the accountcode for the call is set in accordance with the business rules. For example if two customers A and B are talking to each other and A blind transfers B to a billable destination Z then who pays for the call from B to Z? There is no right answer but as far as the CDRs are concerned it's irrelvant as long as each call is recorded and the accountcode can be set within the dialplan both choices can be accomodated. Only if the dial plan actually gets enough information to set the accountcode, which at least historically wasn't the case for Asterisk. In 1.2.x, you couldn't in the dialplan tell if a call went A-B or A-C(SIP redirect)-B. BLINDXFER didn't get set correctly in all cases. The alternative is to use the built-in accountcode from sip.conf; I haven't verified how well that actually works. It won't work if you need to distinguish two different phones behind a SIP trunk, but I don't think anything can, so we can forget about that case. /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] compare Linksys SPA8000 and Grandstream GXW4008
Hi, Has anyone compared SPA8000 vs. GXW4008 especially in terms of firmware and hardware stability (the feature sets are apparently similar)? Vieri ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CDR Rewrite -- Questions to the users
On Sat, Jan 17, 2009 at 10:39 AM, Benny Amorsen benny+use...@amorsen.dk wrote: Only if the dial plan actually gets enough information to set the accountcode, which at least historically wasn't the case for Asterisk. In 1.2.x, you couldn't in the dialplan tell if a call went A-B or A-C(SIP redirect)-B. BLINDXFER didn't get set correctly in all cases. The alternative is to use the built-in accountcode from sip.conf; I haven't verified how well that actually works. It won't work if you need to distinguish two different phones behind a SIP trunk, but I don't think anything can, so we can forget about that case. I've always set the accountcode directly in the dialplan using SetAccountCode and now the newer CDR function. I to encountered occassional problems relying on Asterisk picking up the accountcode from configuration files or a realtime database. We changed our approach to doing a FastAGI call to get the accountcode, the FastAGI call provides the channel name from which the authenticated username and then accountcode can be looked up. As for blind transfers I've always seen the accountcode on the transferred call leg set to that of the call that initiated it. If you wanted it the other way around you do have the option of breaking back into the dialplan when a blind transfer occurs by using the TRANSFER_CONTEXT. At the moment depending on which Asterisk version you are using that won't completely solve the problem since the CDRs produced when transfers occur are all wrong and differently wrong in the different Asterisk versions. Regards, Greyman. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Upgrade
after doing that (erasing Asterisk 1.4 completely and installing Asterisk 1.2) will this impact all of the trunks configurations that are existed in FreePBX that i made before i mean, will i need to make something to operate all these trunks configurations as before?. From: torinti...@hotmail.com To: asterisk-users@lists.digium.com Date: Fri, 16 Jan 2009 16:11:39 +0200 Subject: Re: [asterisk-users] Asterisk Upgrade Thanks to you. Date: Fri, 16 Jan 2009 13:24:16 + From: gordon+aster...@drogon.net To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk Upgrade On Fri, 16 Jan 2009, Alex Balashov wrote: 1. rm -rf /var/lib/asterisk /var/spool/asterisk /etc/asterisk I'd suggest not removing /etc/asterisk if that's the only source of your config files... If you (re)generate them from elsewhere, it's probably OK. and the important one, I'd have thought is /usr/lib/asterisk/modules Gordon 2. Install 1.2.29. Torintino T wrote: How can i erase asterisk 1.4 completely to reinstall 1.2.29 successfully again in steps please. From: torinti...@hotmail.com To: asterisk-users@lists.digium.com Date: Fri, 16 Jan 2009 03:25:33 +0200 Subject: [asterisk-users] Asterisk Upgrade I was Using freepbx-2.1.1, Asterisk 1.2.29 and Asterisk Addons 1.2.9 i tried to upgrade to Asterisk 1.4.22.1 and Addons 1.4.7 all of the IAX trunks got not working at all. I tried to downgrade by make clean; make; make install in Atserisk 1.2.29 directory.but make gives errors in the end. How can i downgrade asterisk again and undo all changes i made?. (in steps please). and can Backup and Restore return all the previous asterisk configurations?. Thanks. Invite your mail contacts to join your friends list with Windows Live Spaces. It's easy! Try it! http://spaces.live.com/spacesapi.aspx?wx_action=createwx_url=/friends.aspxmkt=en-us See all the ways you can stay connected to friends and family http://www.microsoft.com/windows/windowslive/default.aspx ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (678) 237-1775 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Invite your mail contacts to join your friends list with Windows Live Spaces. It's easy! Try it! _ Show them the way! Add maps and directions to your party invites. http://www.microsoft.com/windows/windowslive/events.aspx___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to transfer a call from one AsteriskServerto another
and canreinvite=yes ? 2009/1/17 Lenz Emilitri lenz.lo...@gmail.com Are you sure that the TRANSFER is supported by the other side at all? see http://thread.gmane.org/gmane.comp.telephony.pbx.asterisk.devel/15267 Thanks l. 2009/1/16 Paul bulkm...@monafamily.com Yes, this is the first method I tried. The transfer only works if it is done before a media path is set up to the first box (not answered by the IVR). If it is answered then transferred, I get a 500 internal server error back from the ITSP and the call dies. I never see anything hit the second box. -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Lenz Emilitri *Sent:* Friday, January 16, 2009 10:09 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] How to transfer a call from one AsteriskServerto another I guess you already tried this? http://www.voip-info.org/wiki-Asterisk+cmd+Transfer Thanks l. 2009/1/16 Paul bulkm...@monafamily.com I do have it functioning with Dial(). I was looking for a way to completely move the call from the first box though. When using Dial() media moves, but the call is still tied to the first box. In looking at captures when the call is ended, the first box invites out to the ITSP again, then after receiving a 200ok sends a bye. Also while testing, once the call was up on the second box, I stopped Asterisk on the first box which kills the call. -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Lenz Emilitri *Sent:* Friday, January 16, 2009 12:17 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] How to transfer a call from one AsteriskServer to another Why don't you simply Dial() the call to a separate box keeping Asterisk out of the audio path? l. 2009/1/16 Paul bulkm...@monafamily.com Can anyone tell me how I can completely move an established call off of one Asterisk server to another? In our case we have a server with our IVR. Depending upon digits entered, the call can be transferred to any of our other servers depending where the extension or queue reside. We would like to completely move the call off of the first box so we don't tie up resources on it. In our lab we are testing with 1.4.22.1 Our provider which delivers inbound calls to us uses a Sonus gateway. So far, testing has shown that if we transfer the inbound call prior to any media playback, it works. But, if the IVR plays media, then it is failing, with a 500 internal server error being returned. Thanks for any help ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Loway - home of QueueMetrics - http://queuemetrics.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Loway - home of QueueMetrics - http://queuemetrics.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Loway - home of QueueMetrics - http://queuemetrics.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] pstn hangs up: MWI no message waiting ??
Tilghman Lesher wrote: On Friday 16 January 2009 20:27:57 sean darcy wrote: Tilghman Lesher wrote: On Friday 16 January 2009 17:43:21 sean darcy wrote: Danny Nicholas wrote: Why not do a zap restart instead of restarting asterisk? You could write an AGI to do the ZR when the condition occurred and lines where empty. Yes, a cron job to restart zaptel would cut off any call then existing. But how would I test for it? I can imagine: exten=s,n,ExecIf(some damn thing, System(service dahdi restart)) It's the some damn thing I can't imagine. How do you test if dahdi is acting up? Not a service restart, but a dahdi restart. You can't restart the dahdi service without first stopping Asterisk, anyway. if [ `../asterisk-trunk/contrib/scripts/astcli core show channels | wc -l` = 3 ]; then asterisk -rx 'dahdi restart'; fi Wow. I'll try that tomorrow. Put it as the cmd right after answer(), right? Or maybe, h,1 ? Well anyway, at least I'll be able to receive calls over pstn with dahdi. No, I'd actually recommend that as a cron job. It's basically, restart if idle. Any possibility of actually fixing dahdi? sean ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] pstn hangs up: MWI no message waiting ??
On Saturday 17 January 2009 11:04:33 sean darcy wrote: Tilghman Lesher wrote: On Friday 16 January 2009 20:27:57 sean darcy wrote: Tilghman Lesher wrote: On Friday 16 January 2009 17:43:21 sean darcy wrote: Danny Nicholas wrote: Why not do a zap restart instead of restarting asterisk? You could write an AGI to do the ZR when the condition occurred and lines where empty. Yes, a cron job to restart zaptel would cut off any call then existing. But how would I test for it? I can imagine: exten=s,n,ExecIf(some damn thing, System(service dahdi restart)) It's the some damn thing I can't imagine. How do you test if dahdi is acting up? Not a service restart, but a dahdi restart. You can't restart the dahdi service without first stopping Asterisk, anyway. if [ `../asterisk-trunk/contrib/scripts/astcli core show channels | wc -l` = 3 ]; then asterisk -rx 'dahdi restart'; fi Wow. I'll try that tomorrow. Put it as the cmd right after answer(), right? Or maybe, h,1 ? Well anyway, at least I'll be able to receive calls over pstn with dahdi. No, I'd actually recommend that as a cron job. It's basically, restart if idle. Any possibility of actually fixing dahdi? One thing I would suggest is using the TDM410, instead of the TDM400. The problem may be hardware related. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.6 T38 to G711 transcoding is this possible?
The scenario we have is fax send/recieve software that ONLY talks T38 and an asterisk box. We have ITSP providers that do NOT talk T38 but G711 only. Does asterisk have the capability to take the T38 call from an ATA or T38 software then bridge/transcode it and do G711 out to the PSTN providers? If not is there another product PAID or FREE software or hardware that can do this easily and reliably? Thanks very much! Steve Gladden -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call file in the future
Hello,
I read a thread on the asterisk dev list (call file handling suggestion)
May i have some comment/opinion on these two ways below to place a call file in
the future ? (from the wiki and the asterisk book but added typos and stupidity
come from me)
The best is ? (and should work ?)
tmsp = the delay in future.. say 100 seconds
exten= ra,n,System(NOW='date %S')
exten= ra,n,System(let NOW=$NOW+$tmsp)
exten= ra,n,System(TOUCH_TMSP='date -d 1970-01-01 $NOW sec GMT+1
+%Y%m%d%H%M. %S)
*
or this way ?
exten= ra,n,Set(touchtime=$[${EPOCH} + ${tmsp}])
exten= ra,n,Set(TOUCH_TMSP=${STRFM(${touchtime},GMT+1,%C%y%m%d%H%M%S)
*
next step:
exten= ra,n,System(touch -t $TOUCH_TMSP /tmp/${idclient}.call))
exten= ra,n,System(mv /tmp/${idclient}.call /var/spool/asterisk/outgoing)
Thanks for your attention, happy 2009. and perhaps a reply ?
___
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Re: [asterisk-users] pstn hangs up: MWI no message waiting ??
- sean darcy seandar...@gmail.com wrote: Doug Bailey wrote: - sean darcy seandar...@gmail.com wrote: pstn incoming on a TDM400P, sometimes i* won't answer, going into a loop like this: -- Starting simple switch on 'DAHDI/4-1' [Jan 16 10:38:55] NOTICE[5808]: chan_dahdi.c:7130 ss_thread: Got event 18 (Ring Begin)... [Jan 16 10:38:57] NOTICE[5808]: chan_dahdi.c:7130 ss_thread: Got event 2 (Ring/Answered)... [Jan 16 10:38:57] NOTICE[5808]: chan_dahdi.c:7299 ss_thread: MWI: Channel 4 no message waiting! As part of the implementation of issue 8587, a check was incldued for MWI messages preceded by Ring Pulse Alert Signals (RPAS). The RPAS is answered by chan_dahdi as a standard call and the MWI message is processed. As part of the implementation, if the MWI message was included, the channel was hung up. This did not take into account that possibility of MWI messages included into the to standard CID spills. I believe this is the case here and the MWI portion of the CID spill is causing the channel to hang up. You can look at commit 169154 for a fix or simply remove the ast_hangup calls immediately after the message MWI: channel %d no message waiting!\n and MWI: Channel %d no message w ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call file in the future
On Sat, Jan 17, 2009 at 7:52 PM, didier.cuffaut didier.cuff...@neuf.fr wrote: May i have some comment/opinion on these two ways below to place a call file in the future ? (from the wiki and the asterisk book but added typos and stupidity come from me) The best is ? (and should work ?) This is just me, but if I were going to program calls in the future I would just name them with the time (2009-01-17-20-08.call for four minutes from now, for example) and put them in a directory. The I'd have a cron job running that looked once per minute in that dir and did the mv if found file with that name. Does that make sense? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk support for multiply (two) console dsp devices
Is it possible for asterisk to support multiple USB audio devices independently as Console/Dsp??? At the moment a call comes in and routes to Console/Dsp which is the sound card on the motherboard. What if I needed another audio device so I added a second or third USB audio device. How do I set up asterisk to route sound to the desired card? Thanks, Jerry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call file in the future
On Sat, 17 Jan 2009, randulo wrote: On Sat, Jan 17, 2009 at 7:52 PM, didier.cuffaut didier.cuff...@neuf.fr wrote: May i have some comment/opinion on these two ways below to place a call file in the future ? (from the wiki and the asterisk book but added typos and stupidity come from me) The best is ? (and should work ?) This is just me, but if I were going to program calls in the future I would just name them with the time (2009-01-17-20-08.call for four minutes from now, for example) and put them in a directory. The I'd have a cron job running that looked once per minute in that dir and did the mv if found file with that name. Does that make sense? Not to me. Cron jobs can be delayed, servers can be rebooted and you're suggesting a solution for a problem that already has a solution - ie. set the access time of the file in the future which is what didier.cuffaut is trying to do. For didier.cuffaut: the Asterisk command System() calls the system routine system() which will fork a shell which will then fork to execute your command. It's more efficient to put all the commands in one script and then use System() to execute that file. System(/path/to/script 100) Although personally, I'd probably write a C (or php or perl) program to make things as efficient as possible than use a shell script (or shell commands), to avoid all the forking, but maybe that's just me. Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Sip Trunk registration
Hi Can anybody help me on this ? I am using Asterisknow 1.5.0-Beta(Freepbx) I am having a problem getting the sip trunks to register. It makes no different which provider one is using. Trunk name: callcentric Peer Details: context=from-pstn fromdomain=callcentric.com fromuser=1777xxx host=callcentric.com insecure=very secret=pasword type=peer username=1777xxx Register String: 1777xxx:passw...@callcentric.com/1777xxx I believe that the above configuration is correct. I have tried different sip trunk providers but it makes no difference. They are unable to login. If I put the configuration in to an ata the there on problem. Thanks and regards Andrew ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] canreinvite per route
Can I activate/deactive the canreinvite SIP flag on the dial plan? The idea is to allow reinvite only for exten - exten calls, and not for outbound calls ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sip Trunk registration
core set vervose 100 reload and see what error throw the terminal. 2009/1/17 Zamtron Spain zamt...@terra.es Hi Can anybody help me on this ? I am using Asterisknow 1.5.0-Beta(Freepbx) I am having a problem getting the sip trunks to register. It makes no different which provider one is using. Trunk name: callcentric Peer Details: context=from-pstn fromdomain=callcentric.com fromuser=1777xxx host=callcentric.com insecure=very secret=pasword type=peer username=1777xxx Register String: 1777xxx:passw...@callcentric.com/1777xxx I believe that the above configuration is correct. I have tried different sip trunk providers but it makes no difference. They are unable to login. If I put the configuration in to an ata the there on problem. Thanks and regards Andrew ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ast_yyerror()
Hi All,
I got this error:
[Jan 18 09:56:58] WARNING[9617]: ast_expr2.fl:407 ast_yyerror:
ast_yyerror(): syntax error: syntax error, unexpected $end, expecting
'-' or '!' or '(' or 'token'; Input:
1
^
exten = s,3,GotoIf($[${GROUP_COUNT(${ARG1})} ${LINELIMIT}]?101)
exten = s,3,GotoIf($[${GROUP_COUNT(${TRUNKGROUP})} ${LINELIMIT}]?101)
exten = s,3,GotoIf($[${GROUP_COUNT(${TRUNKGROUP})} ${LINELIMIT}]?101)
not sure where that error is based on the conditions on my dial plan. TIA
Regards,
Ron
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Re: [asterisk-users] canreinvite per route
Have canreinvite set for your internal extens. You can also have canreinvite enabled by default for all and use one or more of the 't','T','h','H','w','W' or 'L' options set in your dial commands which will override the canreinvite option and not send re-invites. cheers - Ben --- On Sat, 1/17/09, Gabriel Ortiz Lour ortiz.ad...@gmail.com wrote: From: Gabriel Ortiz Lour ortiz.ad...@gmail.com Subject: [asterisk-users] canreinvite per route To: asterisk-users@lists.digium.com Date: Saturday, January 17, 2009, 10:06 PM Can I activate/deactive the canreinvite SIP flag on the dial plan? The idea is to allow reinvite only for exten - exten calls, and not for outbound calls ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] pstn hangs up: MWI no message waiting ??
Doug Bailey wrote: - sean darcy seandar...@gmail.com wrote: Doug Bailey wrote: - sean darcy seandar...@gmail.com wrote: pstn incoming on a TDM400P, sometimes i* won't answer, going into a loop like this: -- Starting simple switch on 'DAHDI/4-1' [Jan 16 10:38:55] NOTICE[5808]: chan_dahdi.c:7130 ss_thread: Got event 18 (Ring Begin)... [Jan 16 10:38:57] NOTICE[5808]: chan_dahdi.c:7130 ss_thread: Got event 2 (Ring/Answered)... [Jan 16 10:38:57] NOTICE[5808]: chan_dahdi.c:7299 ss_thread: MWI: Channel 4 no message waiting! As part of the implementation of issue 8587, a check was incldued for MWI messages preceded by Ring Pulse Alert Signals (RPAS). The RPAS is answered by chan_dahdi as a standard call and the MWI message is processed. As part of the implementation, if the MWI message was included, the channel was hung up. This did not take into account that possibility of MWI messages included into the to standard CID spills. I believe this is the case here and the MWI portion of the CID spill is causing the channel to hang up. You can look at commit 169154 for a fix or simply remove the ast_hangup calls immediately after the message MWI: channel %d no message waiting!\n and MWI: Channel %d no message w Thanks. That's a lot better idea than calling Digium Monday and yelling bloody murder. Why in the world would they screw up and obsolete their own hardware? sean ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] pstn hangs up: MWI no message waiting ??
sean darcy wrote: Doug Bailey wrote: - sean darcy seandar...@gmail.com wrote: Doug Bailey wrote: - sean darcy seandar...@gmail.com wrote: pstn incoming on a TDM400P, sometimes i* won't answer, going into a loop like this: -- Starting simple switch on 'DAHDI/4-1' [Jan 16 10:38:55] NOTICE[5808]: chan_dahdi.c:7130 ss_thread: Got event 18 (Ring Begin)... [Jan 16 10:38:57] NOTICE[5808]: chan_dahdi.c:7130 ss_thread: Got event 2 (Ring/Answered)... [Jan 16 10:38:57] NOTICE[5808]: chan_dahdi.c:7299 ss_thread: MWI: Channel 4 no message waiting! As part of the implementation of issue 8587, a check was incldued for MWI messages preceded by Ring Pulse Alert Signals (RPAS). The RPAS is answered by chan_dahdi as a standard call and the MWI message is processed. As part of the implementation, if the MWI message was included, the channel was hung up. This did not take into account that possibility of MWI messages included into the to standard CID spills. I believe this is the case here and the MWI portion of the CID spill is causing the channel to hang up. You can look at commit 169154 for a fix or simply remove the ast_hangup calls immediately after the message MWI: channel %d no message waiting!\n and MWI: Channel %d no message w Thanks. That's a lot better idea than calling Digium Monday and yelling bloody murder. Why in the world would they screw up and obsolete their own hardware? sean OK. Calmer now. If fact a 410 would have the same problem. I'll make the fix on our machines. Should I file a bug, or does the 169154 commit already fix it? sean ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call file in the future
On Saturday 17 January 2009 13:49:16 Gordon Henderson wrote: On Sat, 17 Jan 2009, randulo wrote: On Sat, Jan 17, 2009 at 7:52 PM, didier.cuffaut didier.cuff...@neuf.fr wrote: May i have some comment/opinion on these two ways below to place a call file in the future ? (from the wiki and the asterisk book but added typos and stupidity come from me) The best is ? (and should work ?) This is just me, but if I were going to program calls in the future I would just name them with the time (2009-01-17-20-08.call for four minutes from now, for example) and put them in a directory. The I'd have a cron job running that looked once per minute in that dir and did the mv if found file with that name. Does that make sense? Not to me. Cron jobs can be delayed, servers can be rebooted and you're suggesting a solution for a problem that already has a solution - ie. set the access time of the file in the future which is what didier.cuffaut is trying to do. Almost. It's actually the modified timestamp, not the access timestamp. You cannot usually alter the access timestamp on filesystems, as this is part of the security audit trail. You can use the touch(1) system utility to set the modified timestamp to whatever you like. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] caller ID - handle_request_invite: Failed to authenticate user
We have a caller ID from our phone provider Shaw Cable (digital phone) and it was working OK until recently. I get an error: WARNING[6769]: chan_sip.c:8553 check_auth: username mismatch, have 4, digest has pstn- NOTICE[6769]: chan_sip.c:14316 handle_request_invite: Failed to authenticate user THELMA sip:7804789...@10.10.0.103;tag=50e17675d59121c4o1 at this point call fails, it is not being passed through to asterisk. I'm using Linksys 3102, PSTN answer delay is set to 3sec. to allow for caller ID to pass through. When I decrease timing to 1sec. or eliminate it 0sec the call goes through but there is no caller ID being forwarded. It was working OK for a while. So I'm not sure if Shaw Cable have upgraded something on their digital phone or there is a problem with asterisk/ 4 is a Line1 pstn- is PSTN Line -- #Joseph GPG KeyID: ED0E1FB7 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to transfer a call from one AsteriskServerto another
Who sends the 500 failure code? Asterisk or the VOIP supplier through which you got the call? Note that Asterisk has the basic mechanism for call trasnsfer, just as you transfer a call, so the problem is either in using Transfer() inside IVR context, or the provider. As David noted - use canreinvite=yes And last word: If you get the error from the local Asterisk then raise the verbosity level - probably you'll find some hint there. Good luck, __Yehavi: 2009/1/17 David fire ddf...@gmail.com and canreinvite=yes ? 2009/1/17 Lenz Emilitri lenz.lo...@gmail.com Are you sure that the TRANSFER is supported by the other side at all? see http://thread.gmane.org/gmane.comp.telephony.pbx.asterisk.devel/15267 Thanks l. 2009/1/16 Paul bulkm...@monafamily.com Yes, this is the first method I tried. The transfer only works if it is done before a media path is set up to the first box (not answered by the IVR). If it is answered then transferred, I get a 500 internal server error back from the ITSP and the call dies. I never see anything hit the second box. -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Lenz Emilitri *Sent:* Friday, January 16, 2009 10:09 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] How to transfer a call from one AsteriskServerto another I guess you already tried this? http://www.voip-info.org/wiki-Asterisk+cmd+Transfer Thanks l. 2009/1/16 Paul bulkm...@monafamily.com I do have it functioning with Dial(). I was looking for a way to completely move the call from the first box though. When using Dial() media moves, but the call is still tied to the first box. In looking at captures when the call is ended, the first box invites out to the ITSP again, then after receiving a 200ok sends a bye. Also while testing, once the call was up on the second box, I stopped Asterisk on the first box which kills the call. -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Lenz Emilitri *Sent:* Friday, January 16, 2009 12:17 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] How to transfer a call from one AsteriskServer to another Why don't you simply Dial() the call to a separate box keeping Asterisk out of the audio path? l. 2009/1/16 Paul bulkm...@monafamily.com Can anyone tell me how I can completely move an established call off of one Asterisk server to another? In our case we have a server with our IVR. Depending upon digits entered, the call can be transferred to any of our other servers depending where the extension or queue reside. We would like to completely move the call off of the first box so we don't tie up resources on it. In our lab we are testing with 1.4.22.1 Our provider which delivers inbound calls to us uses a Sonus gateway. So far, testing has shown that if we transfer the inbound call prior to any media playback, it works. But, if the IVR plays media, then it is failing, with a 500 internal server error being returned. Thanks for any help ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Loway - home of QueueMetrics - http://queuemetrics.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Loway - home of QueueMetrics - http://queuemetrics.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Loway - home of QueueMetrics - http://queuemetrics.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by
