Re: [asterisk-users] Sangoma D100 Transcoder Asterisk 1.6
On Mon, Jun 4, 2012 at 4:11 PM, Tim Nelson tnel...@rockbochs.com wrote: - Original Message - I have installed and configures this card in asterisk 1.6. When trying to load the module codec_sangoma.so I see the following in the asterisk log. [2012-06-04 15:50:31] WARNING[18168] loader.c: Error loading module 'codec_sangoma.so': /usr/lib/asterisk/modules/codec_sangoma.so: undefined symbol: ast_config_load [2012-06-04 15:50:31] WARNING[18168] loader.c: Module 'codec_sangoma.so' could not be loaded. Has anyone had a similar issue with this card or have any idea what the undefined symbol: ast_config_load might mean to figure out what direction to head for further debugging? It looks like maybe Wanpipe was not compiled against the same version of Asterisk/DAHDI you're running. That would be the first thing to check. Next stop, Sangoma support. They are fantastic, and support is free. This is all correct, except that codec_sangoma does not come with Wanpipe. Make sure you downloaded the latest version of the sngtc software (the transcoder package) at http://wiki.sangoma.com/Sangoma-Media-Transcoding If you compiled the codec for one Asterisk version and then re-installed another Asterisk version without recompiling sngtc, you may get this issue. *Moises Silva **Manager, Software Engineering*** msi...@sangoma.com Sangoma Technologies 100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada t. +1 800 388 2475 (N. America) t. +1 905 474 1990 x128 f. +1 905 474 9223 **http://www.sangoma.com/contact?utm_source=signatureutm_medium=emailutm_campaign=email+signatures Productshttp://sangoma.com/products?utm_source=signatureutm_medium=emailutm_campaign=email%2Bsignatures | Solutionshttp://sangoma.com/solutions?utm_source=signatureutm_medium=emailutm_campaign=email%2Bsignatures | Eventshttp://sangoma.com/about_us/events?utm_source=signatureutm_medium=emailutm_campaign=email%2Bsignatures | Contacthttp://www.sangoma.com/contact?utm_source=signatureutm_medium=emailutm_campaign=email%2Bsignatures | Wikihttp://wiki.sangoma.com/?utm_source=signatureutm_medium=emailutm_campaign=email%2Bsignatures | Facebookhttp://www.facebook.com/pages/Sangoma-VoIP-Cards/43578453335?utm_source=signatureutm_medium=emailutm_campaign=email%2Bsignatures | Twitterhttp://www.twitter.com/sangoma?utm_source=signatureutm_medium=emailutm_campaign=email%2Bsignatures`| | YouTubehttp://www.youtube.com/sangomatechnologies?utm_source=signatureutm_medium=emailutm_campaign=email%2Bsignatures -- _ -- 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] Sangoma Card Issue
On Wed, May 30, 2012 at 2:34 PM, Eric Wieling ewiel...@nyigc.com wrote: Has anyone experienced an issue with Sangoma analog cards where the card suddenly stops working? Trying to dial out shows the channel as busy, even though there is no active call on that port? I'd like to see the output of wanpipemon -i interface -c astats -m module You should run that command when chan_dahdi is reporting the channel as busy. If you have not contacted techd...@sangoma.com I very much advice you to, that way we can take care of you and get a clear history of your issues on our tracker. *Moises Silva **Manager, Software Engineering*** msi...@sangoma.com Sangoma Technologies 100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada t. +1 800 388 2475 (N. America) t. +1 905 474 1990 x128 f. +1 905 474 9223 **http://www.sangoma.com/contact?utm_source=signatureutm_medium=emailutm_campaign=email+signatures Productshttp://sangoma.com/products?utm_source=signatureutm_medium=emailutm_campaign=email%2Bsignatures | Solutionshttp://sangoma.com/solutions?utm_source=signatureutm_medium=emailutm_campaign=email%2Bsignatures | Eventshttp://sangoma.com/about_us/events?utm_source=signatureutm_medium=emailutm_campaign=email%2Bsignatures | Contacthttp://www.sangoma.com/contact?utm_source=signatureutm_medium=emailutm_campaign=email%2Bsignatures | Wikihttp://wiki.sangoma.com/?utm_source=signatureutm_medium=emailutm_campaign=email%2Bsignatures | Facebookhttp://www.facebook.com/pages/Sangoma-VoIP-Cards/43578453335?utm_source=signatureutm_medium=emailutm_campaign=email%2Bsignatures | Twitterhttp://www.twitter.com/sangoma?utm_source=signatureutm_medium=emailutm_campaign=email%2Bsignatures`| | YouTubehttp://www.youtube.com/sangomatechnologies?utm_source=signatureutm_medium=emailutm_campaign=email%2Bsignatures -- _ -- 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] Telephony Card: GSM slots + Analoge
On Sun, Apr 1, 2012 at 9:04 AM, bilal ghayyad bilmar...@yahoo.com wrote: Dears; I am looking to get a telephony card that has GSM slots (ability to place my GSM card into it) in addition to analoge FXS and FXO. There is a card that I found it but really I do not know how much it is reliable: http://www.atcom.cn/AX2G4A.html Did anyone tried atcom? Is there a similar cards like it (but to be USA, Canada or Europe manufacturing)? There is nothing hybrid like that (GSM + Analog) in the NorthAmerica or Europe to my knowledge. We at Sangoma (from Canada) have a 4-port GSM card though which uses chan_dahdi (patching needed at the moment). *Moises Silva **Manager, Software Engineering*** msi...@sangoma.com Sangoma Technologies 100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada t. +1 800 388 2475 (N. America) t. +1 905 474 1990 x128 f. +1 905 474 9223 **http://www.sangoma.com/contact?utm_source=signatureutm_medium=emailutm_campaign=email+signatures Productshttp://sangoma.com/products?utm_source=signatureutm_medium=emailutm_campaign=email%2Bsignatures | Solutionshttp://sangoma.com/solutions?utm_source=signatureutm_medium=emailutm_campaign=email%2Bsignatures | Eventshttp://sangoma.com/about_us/events?utm_source=signatureutm_medium=emailutm_campaign=email%2Bsignatures | Contacthttp://www.sangoma.com/contact?utm_source=signatureutm_medium=emailutm_campaign=email%2Bsignatures | Wikihttp://wiki.sangoma.com/?utm_source=signatureutm_medium=emailutm_campaign=email%2Bsignatures | Facebookhttp://www.facebook.com/pages/Sangoma-VoIP-Cards/43578453335?utm_source=signatureutm_medium=emailutm_campaign=email%2Bsignatures | Twitterhttp://www.twitter.com/sangoma?utm_source=signatureutm_medium=emailutm_campaign=email%2Bsignatures`| | YouTubehttp://www.youtube.com/sangomatechnologies?utm_source=signatureutm_medium=emailutm_campaign=email%2Bsignatures -- _ -- 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] Telephony Card: GSM slots + Analoge
On Sat, May 26, 2012 at 2:55 AM, Mitul Limbani mi...@enterux.in wrote: Dear Moises, Does Sangoma manufacture 4 port gsm card? or is that an Openvox card? As I said, Sangoma http://www.sangoma.com/products/telecom_boards/wireless/w400.html -- _ -- 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] MFCR2 Long distance calls not connected
On Wed, Dec 28, 2011 at 3:32 PM, Gilberto Verástegui gilbert...@ti-m.com.mx wrote: Calls to long distance get disconnected before answer. Telco: Alestra Country: Mexico System: Elastix 2.2 Digital Card: Digium TE122 Log: [Dec 28 14:37:44] VERBOSE[4586] pbx.c: -- Executing [+525552622900@default:1] Set(SIP/OCS_TRUNK-01bf, EXT=015552622900) in new stack [Dec 28 14:37:44] VERBOSE[4586] pbx.c: -- Executing [+525552622900@default:2] Dial(SIP/OCS_TRUNK-01bf, DAHDI/g1/015552622900,60) in new stack [Dec 28 14:37:44] VERBOSE[4586] app_dial.c: -- Called DAHDI/g1/015552622900 [Dec 28 14:37:44] DEBUG[4586] chan_dahdi.c: bits changed in chan 1 [Dec 28 14:37:53] DEBUG[4586] chan_dahdi.c: disconnecting MFC/R2 call on chan 1 [Dec 28 14:37:53] DEBUG[4586] chan_dahdi.c: ast cause 0 resulted in openr2 cause 6/Normal Clearing [Dec 28 14:37:53] VERBOSE[4586] chan_dahdi.c: -- Hungup 'DAHDI/1-1' [Dec 28 14:37:53] VERBOSE[4586] pbx.c: == Spawn extension (default, +525552622900, 2) exited non-zero on 'SIP/OCS_TRUNK-01bf' [Dec 28 14:37:53] VERBOSE[9190] chan_dahdi.c: MFC/R2 call end on channel 1 Found this email list, but I think is too old. http://www.mail-archive.com/asterisk-users@lists.digium.com/msg205765.html You would be better off asking this questions in asterisk-r2 mailing list. I will answer the same way that I answered back then. You need to enable protocol debugging. Without protocol debugging there is no way to tell what is happening to the call. Read the sample chan_dahdi.conf included with Asterisk and search for mfcr2 logging options. Having said that, it is possible in international calls you need to specify a different caller category. *Moises Silva **Software Engineer, Development Manager*** msi...@sangoma.com Sangoma Technologies 100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada t. +1 800 388 2475 (N. America) t. +1 905 474 1990 x128 f. +1 905 474 9223 **http://www.sangoma.com/contact?utm_source=signatureutm_medium=emailutm_campaign=email+signatures Productshttp://sangoma.com/products?utm_source=signatureutm_medium=emailutm_campaign=email%2Bsignatures | Solutionshttp://sangoma.com/solutions?utm_source=signatureutm_medium=emailutm_campaign=email%2Bsignatures | Eventshttp://sangoma.com/about_us/events?utm_source=signatureutm_medium=emailutm_campaign=email%2Bsignatures | Contacthttp://www.sangoma.com/contact?utm_source=signatureutm_medium=emailutm_campaign=email%2Bsignatures | Wikihttp://wiki.sangoma.com/?utm_source=signatureutm_medium=emailutm_campaign=email%2Bsignatures | Facebookhttp://www.facebook.com/pages/Sangoma-VoIP-Cards/43578453335?utm_source=signatureutm_medium=emailutm_campaign=email%2Bsignatures | Twitterhttp://www.twitter.com/sangoma?utm_source=signatureutm_medium=emailutm_campaign=email%2Bsignatures`| | YouTubehttp://www.youtube.com/sangomatechnologies?utm_source=signatureutm_medium=emailutm_campaign=email%2Bsignatures -- _ -- 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] Asynchronous AGI Problems (Asterisk 1.8.7.0), ubuntu-server
On Sun, Sep 25, 2011 at 8:26 AM, Mehmet Avcioglu meh...@activecom.net wrote: Actually it doesn't say AGI(async:script) it says AGI(async:agi) and than continues further to setting up an AMI user so the script is executed through the manager interface?? Than it says AGI(agi:async).?? Well most importantly it says Cons of async AGI: It is the most complex method of using AGI to implement. ..:) I have been interested in Async AGI as well and after reading your post looked into the link you provided, seems different than what we immediately think, a background process. Perhaps just start the script normally AGI(script.sh) and than inside it run your background process background-script.sh /dev/null 21 /dev/null or fork a new process, detach, run in background, etc... Hopefully somebody else can point us towards the right direction in setting up a real asterisk asynchronous AGI application. Despite being some shameless self-promotion, I want to point out this post I wrote several years ago explaining the basics: http://www.moythreads.com/wordpress/2007/12/24/asterisk-asynchronous-agi/ Moises Silva Senior Software Engineer, Software Development Manager Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON L3R 9R6 Canada t. 1 905 474 1990 x128 | e. m...@sangoma.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sangoma A400 background noise after a while
On Wed, May 4, 2011 at 1:01 PM, M Shokuie sena...@gmail.com wrote: Dear folks, We have recently installed A400D card with 12 FXO modules, the serer is HP DL180 G6, cards works fine but after a while all the calls get an awful noise, you can not get what each side says. The noise cleares as soon as we restart wanrouter but not asterisk (i mean asterisk restart does not solve). We previsouly confronted this situation with PRI cards but not analogs, wanpipe version is 3.5.18 and zaptel 1.4.12 also tested with recent DAHDI with out any help. ifconfig doesnt show any overruns or errors. Once earlier we had the same problem and come to the conclusion to change the mainboard but this time i got mad as i couldnt change a 3000$ HP server that easy. Is there a way i could get if there is any problem of interrupts, when i check interrupts i could not see any shared interrupts for Snagoma card. Anyhelp would be highly appreciated. -- Hello M Shokuie, This kind of troubleshooting is better addressed by Sangoma technical support staff. You can send an email to techd...@sangoma.com and you will be taken care of. Regards, Moises Silva Senior Software Engineer, Software Development Manager Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON L3R 9R6 Canada t. 1 905 474 1990 x128 | e. m...@sangoma.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How is Libpri developped ?
On Fri, Mar 4, 2011 at 12:50 AM, Olivier oza_4...@yahoo.fr wrote: Hi, Can you explain the main differences between Libpri 1.4.11 and 1.4.12 as both seem to receive additions and patches ? Do they target different asterisk versions ? Can they both be considered as production-ready ? 1.4.12 is just a newer version than 1.4.11 and any released version is as production-ready as can be reasonably be expected AFAIK. Moises Silva Senior Software Engineer Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON L3R 9R6 Canada t. 1 905 474 1990 x128 | e. m...@sangoma.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FAX on PRI to MFCR2
On Fri, Feb 18, 2011 at 3:23 AM, leonimar cape leo_mac...@yahoo.com wrote: Hi, I am having issues sending and receiving fax on my asterisk setup. Currently I have a server that has 2 x E1 TDM cards one is sangoma and the other one is openvox. Both support echo cancellation. One of the e1 is connected to our telco provider via mfcr2 where all our incoming calls originate. On the other end is a pri connection going to HICOM PABX where the local attached to a fax is connected. Fax passing thru this connection are not getting thru and always getting drop. FYI: all of the 8xE1s are currently up with two using mfcr2 and the rest is ISDN pri. My guess is that using 2 different cards from 2 different manufacturers for FAX (which is sensitive to time slips) is probably a bad idea because most likely is not possible to synchronize their clocks. If you were using 2 Sangoma ports you can sync the ports with the TE_REF_CLOCK parameter. Moises Silva Senior Software Engineer Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON L3R 9R6 Canada t. 1 905 474 1990 x128 | e. m...@sangoma.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sangoma wanpipe install error
On Sun, Feb 13, 2011 at 9:25 PM, Roi Stork roi.st...@gmail.com wrote: Here's the messages log. There's a line that says ERROR: Unsupported DS E1 CHIP (00:00) That's pretty bad. Could you post the output of wanrouter hwprobe verbose ? Moises Silva Senior Software Engineer Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON L3R 9R6 Canada t. 1 905 474 1990 x128 | e. m...@sangoma.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sangoma wanpipe install error
As the error suggest, try checking /var/log/messages for possible hints on what went wrong. Make sure you configured the device with wancfg_dahdi script first. Moises Silva Senior Software Engineer Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON L3R 9R6 Canada t. 1 905 474 1990 x128 | e. m...@sangoma.com On Fri, Feb 11, 2011 at 1:47 AM, Roi Stork roi.st...@gmail.com wrote: Trying to install wanpipe 3.5.18. No errors during compile. But when I reach the point where wanpipe and dahdi_cfg is started, I encountered an error. Starting WAN Router... Loading WAN drivers: wanpipe done. Starting up device: wanpipe1 wanconfig: WAN device wanpipe1 driver load failed !! : ioctl(wanpipe1,ROUTER_SETUP) failed: : 22 - Invalid argument Wanpipe driver did not load properly Please check /var/log/wanrouter and /var/log/messages for errors Configuring interfaces: w1g1 w1g1: ERROR while getting interface flags: No such device done. /etc/wanpipe/scripts/start: 7: Syntax error: Bad for loop variable DAHDI_SPANCONFIG failed on span 1: No such device or address (6) Dont know why I still keep getting a 'No such device' error even if the device was detected (Sangoma a104de, setup asked to configure/skip the 4 ports) before the error happened. -- _ -- 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] Sangoma U100 failing every Monday - USB port problem or Wanrouter issue?
On Tue, Dec 28, 2010 at 11:33 AM, Bruce B bruceb...@gmail.com wrote: I appreciate your feedback and let me know what info I can post here that may help resolve the issue (such as output from dmesg or lspci?). Hi Bruce, The following would be useful for starters: 1. cat /etc/wanpipe/*.conf 2. ifconfig -a (from a working and non-working situation) 3. lspci -v and lsusb -v (from a working and non-working situation) 4. wanrouter hwprobe verbose (from a working and non-working situation) 5. /var/log/messages (near the date the problem happened) Moises Silva Senior Software Engineer Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON L3R 9R6 Canada t. 1 905 474 1990 x128 | e. m...@sangoma.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unit of measurement dahdi_monitor
On Thu, Nov 25, 2010 at 11:54 AM, Gustavo Santos gust...@voip.ufrj.brwrote: I am studying about echo cancellation in asterisk and I want to use the numeric information from dahdi_monitor verbose for my research. Unfortunately, I couldn't find anything about the unit of measurement used in this tool. Which unit is used to measure the signal level? dahdi_monitor uses the sample values in L16 format. They are in orders of magnitud of G.711. See tables 5 and 6 of the G.711 spec. In the end, the reference value is the dBm (google that). Moises Silva Senior Software Engineer Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON L3R 9R6 Canada t. 1 905 474 1990 x128 | e. m...@sangoma.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Avoiding deadlock
On Wed, Nov 17, 2010 at 9:56 AM, Vilius Adamkavicius vilius.adamkavic...@invade.net wrote: Hi Chad, Thanks for your suggestions. However I believe decreasing logging, its just like closing your eyes and ignoring what happening behind you, the problem is still there. Also decreased logging will prevent from troubleshooting any other problems in the future. Would you happen to know any potential causes for this message? The problem is you were just told by a Digium engineer who knows the code from many years back that is a debug message and there is nothing to worry about and you insist in believing this is a problem. If you want to know what the message means and why you should not worry you must understand what a lock is, what lock contention is and what a deadlock is. Moises Silva Senior Software Engineer Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON L3R 9R6 Canada t. 1 905 474 1990 x128 | e. m...@sangoma.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No translator path exists for channel type DAHDI (native 76) to 256
On Fri, Oct 1, 2010 at 10:08 AM, Danny Dias ing.diasda...@gmail.com wrote: Hello, We are having issues with a NEW Sangoma A108D: -- Executing [691918...@pbx1:1] Dial(SIP/xtravoip200-009d24b0, DAHDI/g0/691918892|30|m) in new stack [Oct 1 10:04:43] WARNING[14171]: channel.c:3170 ast_request: No translator path exists for channel type DAHDI (native 76) to 256 [Oct 1 10:04:43] WARNING[14171]: app_dial.c:1237 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 58 - Bearer capability not available) == Everyone is busy/congested at this time (1:0/0/1) -- Executing [691918...@pbx1:2] Hangup(SIP/xtravoip200-009d24b0, ) This has nothing to do with the card. Format 256 is G729, it looks to me you're SIP phone is trying to place a call to the PSTN using G729, since the PSTN only supports alaw/ulaw (depending on the type of channel), you need to transcode the call, and you don't have G729 codec (you need to pay for it). Sangoma offers D100 card which can do the transcoding to G729 in Asterisk. But the easiest solution is to switch the codec in your softphone to ulaw/alaw or something that is supported natively by Asterisk without extra licensing required. Moises Silva Senior Software Engineer Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON L3R 9R6 Canada t. 1 905 474 1990 x128 | e. m...@sangoma.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sangoma A108 PCIe V2.0
On Fri, Sep 17, 2010 at 11:22 AM, Nyamul Hassan mnhas...@usa.net wrote: While this is too many eggs in one basket, but can be useful if you have too many E(T)1s say equivalent to a STM1 (OC3) or more. In that case, it would be too many boxes at 8ports / box. Somewhere in the mailing list, Sangoma devs said that they do 32E(T)1 per box on the labs quite frequently, although mostly for load testing. That is correct, that is our typical load test scenario. However, Asterisk is a complex system with many features. Our testing focus on SIP to TDM bridging, meaning the only used applications are Answer() and Dial() with the DAHDI and SIP channel drivers, typically with latest 1.4. Additionally we always compile DAHDI modifying the chunk size to reduce the interrupt load. As far as your question about PCIe 2.0, yes the A108 should work just fine there. Moises Silva Senior Software Engineer Sangoma Technologies Inc. | NEW 100 Renfrew Drive, Suite 100, Markham ON L3R 9R6 Canada t. 1 905 474 1990 x128 | e. m...@sangoma.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sangoma A108 PCIe V2.0
Those modifications are done via regular Sangoma installation with a special option to the Setup script. http://wiki.sangoma.com/wanpipe-linux-asterisk-appendix#zaptel_adjustable_chunk_sz http://www.sangoma.com/assets/docs/misc/2009_10_09_How_to_Reduce_Asterisk_System_Loads.pdf Moises Silva Senior Software Engineer Sangoma Technologies Inc. | NEW 100 Renfrew Drive, Suite 100, Markham ON L3R 9R6 Canada t. 1 905 474 1990 x128 | e. m...@sangoma.com On Sat, Sep 18, 2010 at 9:45 AM, Nyamul Hassan mnhas...@usa.net wrote: Thank you for your info Moises. For those who want to have a high density system, can you provide what modifications to the Dahdi (or anything else) do you make? Regards HASSAN On Sat, Sep 18, 2010 at 19:39, Moises Silva moises.si...@gmail.comwrote: On Fri, Sep 17, 2010 at 11:22 AM, Nyamul Hassan mnhas...@usa.net wrote: While this is too many eggs in one basket, but can be useful if you have too many E(T)1s say equivalent to a STM1 (OC3) or more. In that case, it would be too many boxes at 8ports / box. Somewhere in the mailing list, Sangoma devs said that they do 32E(T)1 per box on the labs quite frequently, although mostly for load testing. That is correct, that is our typical load test scenario. However, Asterisk is a complex system with many features. Our testing focus on SIP to TDM bridging, meaning the only used applications are Answer() and Dial() with the DAHDI and SIP channel drivers, typically with latest 1.4. Additionally we always compile DAHDI modifying the chunk size to reduce the interrupt load. As far as your question about PCIe 2.0, yes the A108 should work just fine there. Moises Silva Senior Software Engineer Sangoma Technologies Inc. | NEW 100 Renfrew Drive, Suite 100, Markham ON L3R 9R6 Canada t. 1 905 474 1990 x128 | e. m...@sangoma.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Echo on Sangoma A400 and background noise
On Wed, Sep 15, 2010 at 6:10 PM, Al lists asteris...@gmail.com wrote: I'm a long time user of Digium carts and stupid me i wanted to give Sangoma a try. We got Sangoma A400 with 6 FXO ports. Asterisk version: 1.4.35 Zaptel version: 1.4.11 Wanpipe version: 3.5.11 we tried to use fxtune but looks like it wont work with Sangoma card, ( please correct me if i'm wrong) Echo is really bad and also we have background noise on all lines. We tried both mg2 and oslec echo canceler. was wondering if you have any experiense with that because Sangoma tech support is not helpfull, just look at their response: Hello, I am sorry you've had this bad experience and I apologize on behalf of Sangoma if we gave you the impression of not caring about your technical issue. To rephrase what tech support meant. Our HWEC is known to provide better results, but in no way that means that we will not look at your issue with the card that does not have HWEC. A senior tech support engineer will be contacting you soon today to follow up on your issue appropriately. Regards, Moises Silva Senior Software Engineer Sangoma Technologies Inc. | NEW 100 Renfrew Drive, Suite 100, Markham ON L3R 9R6 Canada t. 1 905 474 1990 x128 | e. m...@sangoma.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to avoid interruptions with DIGIUM
On Thu, Sep 9, 2010 at 11:08 AM, Danny Dias ing.diasda...@gmail.com wrote: Thanks Kevin, But today i saw a Kernel Panic into my server, for no any apparent reasondoes this parameter could help: pci=routeirq By the way, we are using DELL servers, i've also used Sangoma, and always the same problem Thanks! I'd like to know which problem you had with the Sangoma card as there are no shared interrupt issues we know of. There used to be a problem with some Dell servers though, but that was already fixed some weeks ago. Moises Silva Senior Software Engineer Sangoma Technologies Inc. | NEW 100 Renfrew Drive, Suite 100, Markham ON L3R 9R6 Canada t. 1 905 474 1990 x128 | e. m...@sangoma.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] g729 codec loading
Try disabling SELinux if you have it enabled (unless of course you need it). I seem to remember there is certain compilation flags required (position independent code, -fPIC?) to run with SELinux enabled, may be the codec_g729a.so is not compiled properly to run under such circumstances? Moises Silva Senior Software Engineer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. m...@sangoma.com On Fri, Jul 16, 2010 at 5:57 PM, Adolphe Cher-Aime achera...@gmail.comwrote: Hello Everyone, I've successfully registered my g729a licenses. When i try to load the module from asterisk Cli i got the following error *Error loading module 'codec_g729a.so': /usr/lib/asterisk/modules/codec_g729a.so: cannot restore segment prot after reloc: Permission denied* * loader.c:795 load_resource: Module 'codec_g729a.so' could not be loaded. * * * *I'm running asterisk 1.6.2.9 on CentOs 5.4 * * * *Best regards* * * -- Adolphe CHER-AIME Network Integrator CCNA, CCNA VOICE, Global VSAT Forum Certified (509) 3748-3875 / (509) 3449-4280 -- _ -- 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] problems with unicall
I already replied to you in the asterisk-r2 mailing list. Your lines are blocked, the log is telling you that: [May 12 08:58:43] WARNING[2689]: chan_unicall.c:1034 unicall_call: Make call failed - Blocked The only way you get that is if the line is blocked ( rx ABCD bits are 1101 or equivalent blocked for your country ) or the line is configured only for incoming calls ( not possible since chan_unicall.c hard-codes that parameter to allow calls in both ways ). Moises Silva Senior Software Engineer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. m...@sangoma.com On Wed, May 12, 2010 at 5:03 PM, Marcelo nunes dos santos marcelo7...@gmail.com wrote: Hello, i'm using asterisk 1.4.9 in fedora 7, i was compiled its with this package: libpri-1.4.2 asterisk-1.4.9 spandsp-0.0.4 unicall-0.0.5pre1 libmfcr2-0.0.3 libsupertone-0.0.2 libunicall-0.0.3 zaptel-1.4.4 i'm using a E1 pci card with R2 but they not work, when I start the asterisk its generate this log: [May 12 08:53:24] WARNING[30814] channel.c: No channel type registered for 'Unicall' [May 12 08:53:24] WARNING[30814] app_dial.c: Unable to create channel of type 'Unicall' (cause 66 - Channel not implemented) [May 12 08:54:47] NOTICE[2613] cdr.c: CDR simple logging enabled. [May 12 08:54:47] NOTICE[2613] loader.c: 146 modules will be loaded. [May 12 08:54:49] WARNING[2613] res_smdi.c: No SMDI interfaces are available to listen on, not starting SDMI listener. [May 12 08:54:50] WARNING[2613] chan_sip.c: insecure=very at line 37 is deprecated; use insecure=port,invite instead [May 12 08:54:50] WARNING[2613] chan_zap.c: Unable to specify channel 1: Device or resource busy [May 12 08:54:50] ERROR[2613] chan_zap.c: Unable to open channel 1: Device or resource busy here = 0, tmp-channel = 1, channel = 1 [May 12 08:54:50] ERROR[2613] chan_zap.c: Unable to register channel '1-15,17-31' [May 12 08:54:50] NOTICE[2613] pbx_ael.c: Starting AEL load process. [May 12 08:54:50] NOTICE[2613] pbx_ael.c: AEL load process: calculated config file name '/etc/asterisk/extensions.ael'. [May 12 08:54:50] WARNING[2613] ael.y: File: /etc/asterisk/extensions.ael, Line 112, Cols: 34-34: Warning! The empty context ael-dundi-e164-canonical will be IGNORED! [May 12 08:54:50] WARNING[2613] ael.y: File: /etc/asterisk/extensions.ael, Line 120, Cols: 34-34: Warning! The empty context ael-dundi-e164-customers will be IGNORED! [May 12 08:54:50] WARNING[2613] ael.y: File: /etc/asterisk/extensions.ael, Line 128, Cols: 33-33: Warning! The empty context ael-dundi-e164-via-pstn will be IGNORED! [May 12 08:54:50] NOTICE[2613] pbx_ael.c: AEL load process: parsed config file name '/etc/asterisk/extensions.ael' . [May 12 08:54:50] WARNING[2613] pbx_ael.c: Warning: file /etc/asterisk/extensions.ael, line 141-145: The included context 'ael-dundi-e164-canonical' cannot be found. [May 12 08:54:50] WARNING[2613] pbx_ael.c: Warning: file /etc/asterisk/extensions.ael, line 141-145: The included context 'ael-dundi-e164-customers' cannot be found. [May 12 08:54:50] WARNING[2613] pbx_ael.c: Warning: file /etc/asterisk/extensions.ael, line 141-145: The included context 'ael-dundi-e164-via-pstn' cannot be found. [May 12 08:54:50] WARNING[2613] pbx_ael.c: Warning: file /etc/asterisk/extensions.ael, line 276-283: The included context 'ael-parkedcalls' cannot be found. [May 12 08:54:50] NOTICE[2613] pbx_ael.c: AEL load process: checked config file name '/etc/asterisk/extensions.ael'. [May 12 08:54:50] NOTICE[2613] pbx_ael.c: AEL load process: compiled config file name '/etc/asterisk/extensions.ael'. [May 12 08:54:50] NOTICE[2613] pbx_ael.c: AEL load process: merged config file name '/etc/asterisk/extensions.ael'. [May 12 08:54:50] WARNING[2613] pbx.c: Context 'ael-local' tries includes nonexistent context 'ael-parkedcalls' my file unicall.conf is this: [channels] context=e1-inline usecallerid=yes hidecallerid=no threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 relaxdtmf=yes callgroup=1 pickupgroup=1 immediate=no callerid=asreceived musiconhold=default protocolclass=mfcr2 protocolvariant=br,20,20,20 protocolend=cpe group=1 channel = 1-15 channel = 17-31 and when i do any dial by asterisk return this: The 'dial' command is deprecated and will be removed in a future release. Please use 'console dial' instead. == Console is full duplex -- Executing [...@ext-local:1] Dial(OSS/dsp, Unicall/g1/32719595) in new stack [May 12 08:58:43] WARNING[2689]: chan_unicall.c:1034 unicall_call: Make call failed - Blocked -- Couldn't call g1/32719595 -- Hungup 'UniCall/1-1' == Everyone is busy/congested at this time (0:0/0/0) == Auto fallthrough, channel 'OSS/dsp' status is 'CHANUNAVAIL' can somebody help me? Sorry for my english
Re: [asterisk-users] Embedded IAX
http://downloads.asterisk.org/pub/telephony/libiax/ That package is outdated AFAIK but is a start. You should be able to use chan_iax in Asterisk as a reference to fix libiax and use it for your own purposes. Moises Silva Senior Software Engineer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. m...@sangoma.com On Fri, Apr 30, 2010 at 1:23 PM, Bill Shaw b.s...@comcast.net wrote: Hi All, I've been lurking here for a while now, having only made a couple of posts. I am starting a new hardphone project and was wondering if there is some GPL'ed IAX source that I could start with. I've searched and haven't come up with much beyond iaxClient. While iaxClient does give me a little bit to start with, it looks like it is really intended to be more of a softphone running on a Linux machine, and will take some heavy mods to get it running in an embedded DSP environment. Running something like AstLinux on the DSP along with iaxClient may be a possibility but it seems like an awfully lot of baggage to carry around just to get the IAX part of the project. Any pointers would be greatly appreciated. Best, Bill -- _ -- 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] Using asterisk as avaya definity recordingserver
On Mon, Mar 22, 2010 at 7:56 PM, Rafael Prado Rocchi pr...@practis.com.brwrote: Hi, it's not that simple. It requires deep modification on asterisk and dahdi sources to work the way you want. Why? I must confess I still don't quite understand what he wants, from what I've read the legacy pbx will place a secondary call via ISDN ( did he mean PRI? ) therefore Asterisk will just Record(), what is it that is not so simple about that? -- Moises Silva Senior Software Engineer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. m...@sangoma.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Deadlock in Asterisk 1.4.29.1
If you want to open a bug report the proper place to do it is at http://issues.asterisk.org/ Compile with DEBUG_THREADS and DETECT_DEADLOCKS (see make menuselect compiler flags). -- Moises Silva Senior Software Engineer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. m...@sangoma.com On Fri, Mar 5, 2010 at 6:20 AM, Adrien Lemoine alemo...@legos.fr wrote: Hello, I have previously open a topic on the mailing list about deadlocking on Asterisk 1.2.35. After upgrading to 1.4.29.1 we still experienced the same problem : Mar 5 12:05:56] DEBUG[8647] channel.c: Avoiding initial deadlock for channel '0xb7689840' [Mar 5 12:06:41] DEBUG[7130] channel.c: Avoiding deadlock for channel '0xb7c04788' [Mar 5 12:06:41] DEBUG[7130] channel.c: Avoiding deadlock for channel '0xb7c04788' [Mar 5 12:06:51] DEBUG[8647] channel.c: Avoiding initial deadlock for channel '0xb7660fc0' [Mar 5 12:07:02] DEBUG[8647] channel.c: Avoiding initial deadlock for channel '0xb7671d98' [Mar 5 12:07:07] DEBUG[8647] channel.c: Avoiding initial deadlock for channel '0xb76acb08' [Mar 5 12:07:22] DEBUG[8647] channel.c: Avoiding initial deadlock for channel '0xb76621d0' [Mar 5 12:10:55] DEBUG[8647] channel.c: Avoiding initial deadlock for channel '0xb76a2130' [Mar 5 12:11:44] DEBUG[8647] channel.c: Avoiding initial deadlock for channel '0xb7c04788' [Mar 5 12:12:52] DEBUG[8647] channel.c: Avoiding initial deadlock for channel '0xb7675918' [Mar 5 12:15:11] DEBUG[8647] channel.c: Avoiding initial deadlock for channel '0xb76772b0' [Mar 5 12:15:36] DEBUG[8647] channel.c: Avoiding initial deadlock for channel '0xb76acb08' This happen along the day and resulting in a freeze of Asterisk. I mean that I need to kill -9 the process to be able to restart it. There’s one thing different with Asterisk 1.2.35 : no deadlock AVOIDED in warning level while Asterisk is freezed. Thanks for your help. Regards, Adrien .L -- _ -- 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] E1 R2 Congestion Status
On Tue, Dec 22, 2009 at 9:08 AM, Khaled W Chehab kche...@xplorium.comwrote: I have a 'CONGESTION' Status with R2 protocol. While testing this scenario sip GW--àAsterisk –Digium E1 R2 ProtocolàCisco E1 R2 protocolàsip Gw Find below my error and configuration ,where are the errors in my configuration ? Typically you will be better off in the asterisk-r2 mailing list. Your message is more easily spotted by R2 people. First thing is to check you have green status in your E1, do you? and make sure you have the right clock settings, I never configured a cisco but it seems you configured the cisco to be a slave and I don't see any port in DAHDI system.conf with master clock settings. (span=1,0,0...etc) After that, enable R2 call logging using mfcr2_call_files=yes and pastebin the generated call file (if any), if no file is generated it means the problem is not at R2 but in your local trunk settings (may be dialing in the wrong group or something). -- Moises Silva Senior Software Engineer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. m...@sangoma.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] Problem with sounds DTMF's phone keys
On Mon, Nov 16, 2009 at 3:27 PM, Diana Lopez dlo...@palosanto.com wrote: Hello everybody, I need help, I have a problem with conferences in asterisk, when many people are in a conference sometimes there're users pressing phone keys and this action emits a sound (DTMF of the phone keys), so, I need to find the way of not listening this sound.. I'm using MeetMe(variable,pFX).. I tried whithout F but it doesn't work because users continue listening de DTMF's sounds... But the way, I'm using Asterisk 1.4.26.1 Thanks a lot. This is just a guess, but, I think that if any of the users is using inband DTMF and its sip.conf is not enabled to detect inband dtmf, you may have this result. The 'F' options MUST work if all the peers are properly configured, otherwise is a bug. I order to asterisk to mute or ignore that DTMF it must first properly detect it. Use the /etc/asterisk/logger.conf configuration to display detected dtmf and make sure all users dtmf is being detected. If the dtmf is detected by asterisk and you use the F option in meetme and still you hear the dtmf, then is a bug, try to reproduce it with latest 1.4 version, if you can reproduce it the bug must be filled in issues.asterisk.org. -- Moises Silva Software Developer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. m...@sangoma.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] Async Agi problem
because read=agi lets you read agi events, not send agi actions, agi in write= must be set too if you want to send agi commands. On Mon, Nov 2, 2009 at 3:25 AM, Robert Bielik robert.bie...@xponaut.sewrote: Robert Bielik skrev: Ok, now pretty much everything is up 'n running, however when I try to send an ANSWER (or any) command to *, it replies with org.asteriskjava.manager.response.ManagerError Permission Denied. In manager.conf for the *-java client, I have read = system,call,log,verbose,agent,user,config,dtmf,reporting,cdr,dialplan,agi write = system,call,agent,user,config,command,reporting,originate * is 1.6.1.4 and *-java is 1.0.0 Hmm... setting write = all makes it work... /Rob ___ -- 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 -- Moises Silva Software Developer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. m...@sangoma.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] Async Agi problem
On Fri, Oct 30, 2009 at 4:25 AM, Robert Bielik robert.bie...@xponaut.sewrote: Yeay!! Thank you! No, I have not. And I suspected that I had to put something there, I've googled mad for it but have not found one document saying that I should. Now I know :) Perhaps this is something to be added to the default manager.conf ?? ;) I agree, will get that added to the manager.conf sample -- Moises Silva Software Developer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. m...@sangoma.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] Asterisk-1.6.1.8 DTMF with SIP is not working
On Fri, Oct 30, 2009 at 10:48 PM, Joseph syscon...@gmail.com wrote: On 10/30/09 17:28, Joseph wrote: Trying asterisk-1.6.1.8 but I can not get dtmf signal to go through. dtmfmode = rfc2833 is there still problem with dtmf in version 1.6 ? :-( Will DTMF + sip ever be solved in asterisk or it is impossible task? I remember having problems with DTMF ever since ver. 1.2 :-/ Your question is too general. I don't remember ever having a DTMF problem since 1.0, so it depends on the use you give to asterisk and the equipment you use. -- Moises Silva Software Developer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. m...@sangoma.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] Async Agi problem
On Thu, Oct 29, 2009 at 3:48 AM, Robert Bielik robert.bie...@xponaut.sewrote: and I can see that the context is hit when dialing into *. However my java app that's supposed to receive async agi events get no such events at all, but it does receive other manager API events. * version is 1.6.1.4 You mean you cannot see AsyncAGI events? did you enable agi in the read= parameter in manager.conf for your Java application user? Can you send AGI commands to the channel through the manager? or through the Asterisk CLI agi exec cmd?? -- Moises Silva Software Developer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. m...@sangoma.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] Possible bug in app_meetme.c
On Sat, Oct 17, 2009 at 2:26 PM, Richard Kenner ken...@gnat.com wrote: Is this patch correct? The doesn't make logical sense to me. I think it should be || and making this change fixes the problem I have with SIP phones in MeetMe conferences. If it's correct, is there someplace more formal that I should submit it to? I don't know if it's correct since I don't know what the CONFFLAG_OPTIMIZETALKER flag does nor what your problem is. Giving more background info should help. If you want to submit a patch use issues.asterisk.org and read the guidelines before submitting the bug report. -- Moises Silva Software Developer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. m...@sangoma.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G.729 and Voicemail
I would be very surprised if that were true. Your phones speak many codecs, but they negotiate with asterisk on registration which one they will be using. They don't switch codecs based on the remote channel (which they don't even know about). Today, if your phones are negotiating 729 on registration, you are definitely transcoding calls to/from the PSTN. Assuming we're talking about SIP (and any other voip protocol I know of for that matter), that is incorrect, codec negotiation is done during the call setup based on preferences stored in both Asterisk and the phone (that is what the SDP is for among other things). However at that point Asterisk does not know that the dial plan is going to call the voicemail application to play a file in g729 format (how can possibly now that), and therefore when the file is being played the phone already expects the audio for the call in the format negotiated during call setup which may or may not be g729. Not sure if a re-invite could be issued to change the codec type in the middle of the call, but I suppose it should be possible to implement. -- Moises Silva Software Developer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. m...@sangoma.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] g729 free codec any idea
On Thu, Oct 8, 2009 at 1:33 PM, DHAVAL INDRODIYA dhaval.it01...@gmail.comwrote: there is free implementation of g729 codec you can get it from http://asterisk.hosting.lv I'm not an expert on patents, but even when you have access to the g729 implementation, the algorithm is patented, so, as the disclaimer says in the web page, you still need to pay royalty fees to the g729 patent holders somehow. Unless you live in a country where patents do not matter. -- Moises Silva Software Developer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. m...@sangoma.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] E1/T1 Tapping call recording in Asterisk - Testing needed
On Thu, Oct 1, 2009 at 7:57 AM, DHAVAL INDRODIYA dhaval.it01...@gmail.comwrote: how can i used this patch with digium cards, i have digium card and also having some issue in recording , can you give me procedure for it? May be Martin can help with that, I don't know how to setup Digium boards in high impedance mode. It seems the feature may not be exported via configuration files yet, so changes to the driver may be needed? -- Moises Silva Software Developer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. m...@sangoma.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] E1/T1 Tapping call recording in Asterisk - Testing needed
Howdy, I've spent a couple of days writing a new feature for Asterisk that allows to record calls in T1 or E1 PRI lines using Asterisk connected to tapped lines. This means that you don't have to install anything in the PBX's/telco equipment that is going to be monitored, all you need is to install a device like the PN 633 Tap Connection Adapter that is available for example, from Sangoma, however I am sure there must be other vendors out there offering similar devices. Then you need to pull a pair of cables out of the adapter to your monitoring system with Sangoma boards configured in high impedance mode (I don't know if Digium or other vendors boards expose that functionality to users, but you may want to test and find out if works). More detailed instructions can be found at Sangoma's site or my blog: http://wiki.sangoma.com/sangoma-tap-system http://www.moythreads.com/wordpress/2009/09/26/sangoma-tapping-solution-for-asterisk/ The patches are already out there in the bug tracker along with some SVN branches. https://issues.asterisk.org/view.php?id=15970 https://issues.asterisk.org/view.php?id=15971 I'd love to get feedback in the bug tracker in order to get this feature into Asterisk soon :-) Also don't hesitate in asking for help with the configuration. -- Moises Silva Software Developer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. m...@sangoma.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] E1/T1 Tapping call recording in Asterisk - Testing needed
Is your code vendor locked to Sangoma ??? Hello Martin, not at all. The code is intended to be part of chan_dahdi Asterisk channel driver and as such any card capable of using the dahdi interface can benefit from it. -- Moises Silva Software Developer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. m...@sangoma.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sangoma A200 and battery removal detection ??!!!
On Sat, Sep 19, 2009 at 3:33 AM, M Shokuie sena...@gmail.com wrote: Dear Folks, Anyone knows if Sangoma supports or going to provide support for battery removal detection on FXO lines?? As Tzafrir said earlier DAHDI supports it, which is a very nice feature but what about Sangoma? Regards. -- M. Shokuie Nia. Hello Shokuie, Tzafrir clarified to me that what you were asking for is alarm notification (red alarm / alarm cleared) on battery removal and not hook notifications. Until today, this wasn't implemented in Sangoma Wanpipe drivers, but it was simple enough that I just did the quick fix and this little feature should be available in the next wanpipe release within this week. -- Moises Silva Software Developer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. m...@sangoma.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sangoma A200 and battery removal detection ??!!!
On Sat, Sep 19, 2009 at 3:33 AM, M Shokuie sena...@gmail.com wrote: Dear Folks, Anyone knows if Sangoma supports or going to provide support for battery removal detection on FXO lines?? As Tzafrir said earlier DAHDI supports it, which is a very nice feature but what about Sangoma? Hello M. Shokuie, In Asterisk installations, Sangoma boards use DAHDI drivers to hook up to Asterisk ( because chan_dahdi works only with DAHDI devices). If you were using FreeSWITCH you could use native wanpipe devices /dev/wanpipex_xx instead of /dev/dahdi/x, which uses native Sangoma drivers. In both modes the drivers notify on-hook, off-hook events depending on the battery status. -- Moises Silva Software Developer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. m...@sangoma.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G729
On Thu, Sep 17, 2009 at 10:50 AM, Gordon Henderson gordon+aster...@drogon.net gordon%2baster...@drogon.net wrote: On Thu, 17 Sep 2009, Tilghman Lesher wrote: The free one or the Howlets one? However I can't see how the binary blobs of patented code which digium sells doesn't voilate the GPL either. It's nice to have competition. Keeps you on your toes. Gordon Because Digium OWNS the Asterisk code, and they make an exception for their binary code, is their right as owners (copyright holders) of the code. -- Moises Silva Software Developer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. m...@sangoma.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G729
Close enough. Digium doesn't own the Asterisk code, but it possesses enough of a copyright interest in the code, as well as licenses from all contributors, in order to be able to make that exception. Ah, yeah, I stand corrected. I should have not used own. For the casual reader, the clarification means developers contributing to Asterisk still own the code, but the disclaimer signed by them gives Digium enough rights to make the exception. -- Moises Silva Software Developer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. m...@sangoma.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Platform decision ...
Hi there, I though to chime in here just to share my opinion for what is worth. As a developer who enjoys playing with telephony in general I try to remain as objective as possible when talking about one or the other, and I felt that N from arcdiv was a bit unfair with FreeSWITCH docs. Then you'd have to add the con: cryptic, difficult to find, and wholly incomplete documentation. Don't get me wrong. FreeSwitch is a very nice back-end product. It's hard to not get you wrong when, in my opinion, you start by writing as facts what is barely your particular poor user experience with it. Others, including me, have found what they need in FreeSWITCH wiki just as I have found what I need about Asterisk docs in voip-info. far as ease of putting it into deployment goes, it's a nightmare from its complete dearth of anything related to coherent docs. It still feels very nuts and bolts. Like being handed a Porsche Boxter engine, frame, and a wrench and being told to sort of 'figure out' how it all goes together. And even when you do, it will function screamingly well. But it won't have doors, windows, AC, or creature comforts that we've all come to expect. You mean comforts which you have come to expect. Again, my needs have been so far fulfilled for conferencing and SIP/PSTN gateway uses. Pointing to particular missing applications instead of making your own analogy would be useful, otherwise you are not really being of much help, and just introducing FUD. Many users are confused because they try to do things the same way they are used to with Asterisk and some concepts just don't fit or are differently applied. From what I've seen the users that get annoyed the most are those who keep trying to do things in the Asterisk-way and get overwhelmed by the configuration differences, instead of learning the FreeSWITCH-way to accomplish the same goals. Users just get impatient because they're already familiar with something and this new engine is not managed as the old one. The recent announcement of FreePBX running over FreeSWITCH ( http://www.freepbx.org/news/2009-08-04/freepbx-v3-come-help-us-shape-the-future) should help to close the gap in user configuration and ease of management. Of course, there is some truth in your statements. FreeSWITCH needs to catch up with documentation, but I would defy anyone to say they've come to hang around on IRC and did not get their question answered. Both Asterisk and FreeSWITCH share features, pro's, cont's and for some people one is better than the other. I am looking forward for people to make informed opinions about their experience with both engines. -- Moises Silva Software Developer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. m...@sangoma.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Platform decision ...
possibly very simple needs, but that doesn't make it very complete documentation. Which nobody has stated it is. Just means for your needs (which you have never stated nor pointed to particular missing chunks of documentation for what you intended to do) was far from fulfilling your expectations. I'm not introducing FUD by stating my opinion about the lack of documentation, Moises. You're sounding incredibly defensive. Why? I though I made it clear. Your comment sounded to me like FUD, since you give an opinion without pointing to particular instances of your claims, saying wholly incomplete documentation sounds extreme (which is typical when trying to introduce FUD). So, is just you sounded like that to me, if you are not trying to do so, then don't get it personal, isn't the point here, for me anyways. IRC answers from people hanging around is not documentation. It's what open-source developers like to think of, sometimes, as a 'complete' solution, but it doesn't even come close. Again, never stated IRC was documentation, and was rather introduced as way of saying, what you cannot find on the wiki, you can find it on IRC, because is for sure there are things that are not on the wiki, even for more used technologies like Asterisk. documentation. Which is lacking. This is not FUD. This is not me saying Don't touch FreeSWITCH. This is me saying that, if you're looking for a product with good docs to make an easy transition from traditional PBX tech to new, or even an easy transition from Asterisk to something else, you will not find them with FS. That is at least a bit better than your first post, re-read your first post again, you never even mentioned transition from Asterisk nor whether you wanted a PBX. Even when Asterisk is sometimes seen as a PBX, can be used as a pstn gw or something else, so, as I said, you never mentioned what you wanted to do nor when did you try (as projects evolve, your statements may no longer be true). In short, the only piece of relevant information I was able to take from your initial post was documentation is crap and no useful applications exist, for me, that sounds a lot like FUD. But hey, that's just me, I'm sure others will learn something from your post, I just felt the need to give my opinion as you did. I'm sorry you're offended by my opinions, but in your words, 'I defy you' to show me some comprehensive FreeSWITCH docs. Heck, even SER has more comprehensive documentation, and that's saying a LOT. No offense taken ;-), hope is the same for you. Again, our perspective of what comprehensive documentation is differs, needs improvement for sure though. -- Moises Silva Software Developer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. m...@sangoma.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best ISDN BRI solutions?
On Thu, Aug 6, 2009 at 6:19 AM, Benny Amorsen benny+use...@amorsen.dkbenny%2buse...@amorsen.dk wrote: Moises Silva moises.si...@gmail.com writes: Just for the record, Sangoma Media Gateway does exactly that, leave all your PSTN interfaces (BRI, SS7, PRI) in another box and communicates with Asterisk through the Woomera protocol. What is the advantage of doing Woomera rather than a complete SIP gateway? Is Woomera hardware that much cheaper than e.g. SIP-to-PRI? Is it easier to configure? It can be argued that is easier to configure, but I'll leave that to sysadmins to decide. The point is Woomera is not meant to replace SIP or compete with it, is just an easy way to distribute your TDM interfaces and not have them in the same server where you do media processing, IVR, routing etc. This should leave the picture more clear http://wiki.sangoma.com/wanpipe-linux-asterisk-ss7 So typically you will send all your TDM calls to 1 or more Asterisk servers running chan_woomera and any other service you want to provide in there (including SIP outbound calls). -- Moises Silva Software Developer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. m...@sangoma.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best ISDN BRI solutions?
I might even be willing to try out a more expensive PRI card if I knew it also supported BRI: just as long as I would no longer have to worry about the software support for it -- for both Asterisk 1.4 and 1.6. Thanks, Jaap You can use Sangoma Media Gateway along with Asterisk ( http://wiki.sangoma.com/sangoma-wanpipe-smg-asterisk-bri-installation) That is known to work pretty well for lots of people. -- Moises Silva Software Developer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. m...@sangoma.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best ISDN BRI solutions?
On Wed, Aug 5, 2009 at 5:28 PM, Jorge Mendoza mend...@tcc.com.pe wrote: We use Patton BRI gateways. No problems so far. If possible, we prefer to keep telephony interfaces out of Asterisk box. Regards Jorge Just for the record, Sangoma Media Gateway does exactly that, leave all your PSTN interfaces (BRI, SS7, PRI) in another box and communicates with Asterisk through the Woomera protocol. -- Moises Silva Software Developer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. m...@sangoma.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX for internet file transfer?
On Fri, Jun 26, 2009 at 6:48 PM, Maris maris@vdi.de wrote: I'm dealing with an idea to exchange data in a socket connection style or a sort of ftp transfer with IAX2 as the transport medium. An IAX client on e.g. a notebook could establish a connection to any remote machine (also client) via any Asterisk Server where both clients are registered. Due to the unique properties of IAX2 one could connect quite easily to any hidden remote computer without server functionality and exchange data. To my opinion it should be quite simple to bypass audio-RTP packet conversion in order to allow digital data transmission. Just a question since I am not quite sure I understand your suggestion. How do you plan to reliably transmit a file through UDP which does not guarantee delivery?, not to mention that IAX2 does not use RTP. Are you suggesting to change the protocol to support such transfers? -- Moises Silva Software Developer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. m...@sangoma.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] IAX for internet file transfer?
On Fri, Jun 26, 2009 at 8:26 PM, Martin asteriskl...@callthem.info wrote: I'm sure he meant UDP not RTP. In order to guarantee the delivery you can simply do what IAX already does ... ACK the frames. This is what TCP does and ISDN PRI protocol layer 2 on the T1/E1. But why does he want to do it ? Share secret / illegal files LOL ? Martin I would think IAX ack just the signaling frames, not every single audio frame, does it? -- Moises Silva Software Developer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. m...@sangoma.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] 1000Hz kernel
On Tue, Jun 23, 2009 at 4:15 PM, Danny Nicholas da...@debsinc.com wrote: Yes it will be beneficial. We have a TDM410P and were getting nominal performance until we recompiled the kernel. A good use of 6 hours. Most likely that was just incidental. Having either Digium or Sangoma hardware is enough in order to get the required timing. The only reason to increase the kernel timer to 1khz is when you need dahdi_dummy module, which uses this timer to fake interrupts that otherwise would be generated by real hardware. -- Moises Silva Software Developer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. m...@sangoma.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] How run AsyncAGI commands in background
On Fri, Jun 19, 2009 at 5:32 AM, Jose Ariascyr2...@gmail.com wrote: Hi Moy, I'm using an asterisk 1.4.18 from scratch patched with the last AsyncAGI patch, which fixes a bug about stopping AsyncAGI applications, as may be you can recall from the thread [asterisk-users] async agi question in http://lists.digium.com/pipermail/asterisk-users/2009-April/230488.html. This patched asterisk works fine and it stops the async agi applications launched from the AsyncAGI loop before the Redirect as it's expected. It's for that I don't think stopping the mixmonitor application launched from the AsyncAGI loop would be a bug if I redirect the call. I would be only getting the same behavior than I got with the stream file application as you explained it should be at http://www.moythreads.com/wordpress/2007/12/24/asterisk-asynchronous-agi/#comment-365 I'm only asking if there's any way to prevent stopping applications launched on a channel from the AsyncAGI loop if this channel is redirected afterward, with something like a continue_running_in_background flag in the previous AGI invocation from AMI. Of course, it bring us the problem we'll need some kind of identifier and some stop action to be able to stop those applications running in background launched from the AsyncAGI loop Anyway, as you asked me some days ago, I have published at http://docs.google.com/View?id=ahfnfrcrh3rr_4dkcx9dgw a simple configuration and a simple scenario in order you can try to reproduce what I'm saying. I don't need anyone to do anything for me. I'm willing to do the work, I like programming and trying new things as well, but I'll need some guidelines to go straight ahead. Jose, the thing is that MixMonitor IS a background application in nature, that's why I say is unexpected that after a redirect the recording no longer works. In fact, that's why StopMixMonitor application is needed, because all MixMonitor does is to launch a background thread that hooks into the channel audio, then the channel continues to execute other applications in the dial plan while this background thread monitors its audio, on a redirect StopMixMonitor thread should continue saving audio until StopMixMonitor is called. -- Moises Silva Software Developer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. m...@sangoma.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] Writing for asterisk
this is possibly the best you can find: http://www.russellbryant.net/blog/2008/06/30/how-to-write-an-asterisk-module-part-3/ On Thu, Jun 11, 2009 at 7:50 PM, Carlos Ruiz Diazcarlos.ruizd...@gmail.com wrote: Hello, Where can I found information about writing modules, applications and low level interactions for asterisk? At http://www.asterisk.org/developers I was unable to find tutorials for doing what I mentioned above. Thanks in advance. Carlos. ___ -- 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 -- Moises Silva Software Developer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. m...@sangoma.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] chan_dahdi missing in * 1.6.1.1
On Tue, Jun 9, 2009 at 12:31 AM, Steve Reposcmu...@gmail.com wrote: What is the magic to compile chan_dahdi.so in asterisk 1.6.1.x? I'm on centos 5.3. Also, asterisk 1.4.25 cannot compile chan_dahdi as well while the previous versions do. What changed or what am i missing? There probably isn't magic. If you post the errors you got during the compile we'll be more likely to be able to tell you what's going wrong. Specifically the stuff you got when you said you cannot compile chan_dahdi.so would be important to post. Ah, there are no errors as such during compile or atleast I didn't seem to notice any. However, I do not see chan_dahdi.so in lib/asterisk/modules/ after asterisk install. Attached are some info and config.log from asterisk 1.6.1.1.. # /opt/dahditools/sbin/dahdi_hardware pci::03:04.0 wanpipe- 1923:0040 Sangoma Technologies Corp. A200/Remora FXO/FXS Analog AFT card # /opt/dahditools/sbin/dahdi_scan [1] active=yes alarms=UNCONFIGURED description=wrtdm Board 1 name=WRTDM/0 manufacturer= devicetype= location= basechan=1 totchans=24 irq=0 type=analog port=1,none port=2,none port=3,FXO port=4,FXO port=5,none port=6,none port=7,none port=8,none port=9,none port=10,none port=11,none port=12,none port=13,none port=14,none port=15,none port=16,none port=17,none port=18,none port=19,none port=20,none port=21,none port=22,none port=23,none port=24,none # cat /proc/dahdi/1 Span 1: WRTDM/0 wrtdm Board 1 (MASTER) 1 WRTDM/0/0 2 WRTDM/0/1 3 WRTDM/0/2 4 WRTDM/0/3 5 WRTDM/0/4 6 WRTDM/0/5 7 WRTDM/0/6 8 WRTDM/0/7 9 WRTDM/0/8 10 WRTDM/0/9 11 WRTDM/0/10 12 WRTDM/0/11 13 WRTDM/0/12 14 WRTDM/0/13 15 WRTDM/0/14 16 WRTDM/0/15 17 WRTDM/0/16 18 WRTDM/0/17 19 WRTDM/0/18 20 WRTDM/0/19 21 WRTDM/0/20 22 WRTDM/0/21 23 WRTDM/0/22 24 WRTDM/0/23 I does not matter if i pass --with-dahdi to ./configure script or not. After running ./configure you will get an output file named config.log, that file has the details about which tests were performed during the configuration to determine which headers and libraries were present in the system. You should pastebin that file and post here a link to it. Also, what do you see when you do make menuselect - channel drivers - chan_dahdi? if Asterisk see dahdi, then you should see it marked for compilation. -- Moises Silva Software Developer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. m...@sangoma.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] How run AsyncAGI commands in background
On Sun, Jun 7, 2009 at 4:37 PM, Jose Ariascyr2...@gmail.com wrote: Hi Moy, I'll do it so, but for your answer, it seems you are thinking about it as it could be a bug. I don't think so. I mean: the redirect action on a channel in AsyncAGI stops the current agi execution. It's the normal behavior. It's the way to stop a playfile on a channel if it was previously launched from AsyncAGI: making a redirect out of the AsyncAGI loop. Therefore, when I realized the previously launched EXE MixMonitor AsyncAGI execution was stopping after doing a redirect to meetme, I didn't think it was a bug. I though what I was needing it was a way to tell AsyncAGI, hey, don't stop this agi execution on the channel, even it will be redirected out of AGI on an individual basis for each AsyncAGI EXEC command launched. Thanks Jose The way I see it if you make EXEC MixMonitor inside AsyncAGI loop and then redirect to MeetMe and you don't get the audio recorded, then it's not a normal behavior, MixMonitor is an application that should passively monitor the channel audio independently of where the channel is (regardless of whether the command was executed in Async AGI or dial plan or whatever). However you are also using an old asterisk version and is not likely you can report a bug unless you upgrade to the latest Asterisk and reproduce without a patched Asterisk (for example executing EXEC MixMonitor inside a regular AGI script and then redirect to MeetMe). -- Moises Silva Software Developer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. m...@sangoma.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] Sangoma A200 wt HW echo on 64bit Centos , 4 fxo and crosstalk!
Currently we have put in a temp OpenVOX tdm400 card and it works perfectly. As soon as we swap that and use Sangoma via wanrouter we get crosstalk. For example, if an existing call is happening and a new internal to external call or vise versa happens, they can hear each other, even just to IVR. How often does this happen? (the cross-talk) every single call? is easy to reproduce? Any ideas? All wiring has been checked and this *does not*, I repeat, *does not* happen with the Sangoma card. So what ever explaination we come up with, that fact remains and we get stumped. You meant that this does not happen with the OpenVox card, didn't you? otherwise, you lost me. If you can easily reproduce this, I'd be interested in look into it. -- Moises Silva Software Developer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. m...@sangoma.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] How run AsyncAGI commands in background
then it should work, create a *simple* extensions.conf and pastebin it along with instructions so I can try to reproduce. On Sat, Jun 6, 2009 at 5:02 PM, Jose Ariascyr2...@gmail.com wrote: Hi, Asterisk 1.4.18 AsyncAGI patch from http://moythreads.com/testasync2.diff Regards Jose ___ -- 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 -- Moises Silva Software Developer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. m...@sangoma.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] Sangoma A200 wt HW echo on 64bit Centos , 4 fxo and crosstalk!
On Sat, Jun 6, 2009 at 3:18 PM, Gavin Henrygavin.he...@gmail.com wrote: Every call as soon as the sangoma card is live. Speak to Konrad on your techdesk for more info. Thanks. I'll speak with him on Monday. However if you can provide more information before Monday I will be able to think beforehand on this matter. So please confirm this. If you get an incoming call and send it to Playback(demo-congrats) and then receive a second call and send it to Playback(tt-monkeys), both callers will listen both demo-congrats and tt-monkeys sounds? -- Moises Silva Software Developer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. m...@sangoma.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] How run AsyncAGI commands in background
On Sat, Jun 6, 2009 at 7:18 PM, Philipp Kempgenphilipp.kemp...@amooma.de wrote: Jose Arias schrieb: Hi, Asterisk 1.4.18 AsyncAGI patch from //http://moythreads.com/testasync2.diff http://moythreads.com/testasync2.diff// Regards So what? What do you mean with so what?, if you have not been involved in the conversation you would not understand. http://lists.digium.com/pipermail/asterisk-users/2009-June/232995.html -- Moises Silva Software Developer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. m...@sangoma.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] How run AsyncAGI commands in background
On Fri, Jun 5, 2009 at 4:32 PM, Jose Ariascyr2...@gmail.com wrote:
I'm sorry,
in my last email, where I said redial, I mean redirect.
Thanks
Jose
2009/6/5 Jose Arias cyr2...@gmail.com
Hi all,
I have an external application commanding asterisk by AMI and AsyncAGI. I
also have a dialplan like this:
; AsyncAGI extensions
exten = _8.,1,Noop(entering in AGI loop at 8 ${EXTEN});
exten = _8.,n,AGI(agi:async);
exten = _8.,n,Hangup();
; Meetme extensions
exten = _1.,1,Noop(Conference ${EXTEN} ${CONTEXT});
exten = _1.,n,Set(MEETME_RECORDINGFILE=${EXTEN:}-${UNIQUEID:})
exten = _1.,n,MixMonitor(${MEETME_RECORDINGFILE}.wav)
exten = _1.,n,Meetme(${EXTEN},qdx);
exten = _1.,n,Hangup();
It works fine:
Incoming channels are sent to meetme by an external application, which
receives events by AMI and decides what meetme to use, making a redirect
action to it by AMI. Every channel falling in a meetme (dynamically
created) is recorded by the MixMonitor application.
But there's a little problem:
I don't need to record all calls but only those ones are switable of be
recorded (by some kind of external rules). As it's a waste of cpu and space
to record everything and then to discard almost all of them but some few
ones, I tought to use AsyncAGI to recording only some calls by sending an
AsyncAGI EXE MixMonitor command instead of the dial plan approach.
To do that, the external application, instead of making the redial to
meetme, it must make the redial to an AsyncAGI extension, then it must make
the AGI EXE MixMonitor action, and finally it must make the original
redirect to meetme.
But it doesn't work :-(
When the application reachs the third step (redial to meetme while the
channel is still into the AGI loop, after having sent it the AGI EXE
MixMonitor action) the MixMonitor AGI action is stopped automatically and
the recording ends.
Therefore, does anyone know how to manage that an AsyncAGI action to
remain running in background even if the channel is redirected out of AGI?
Thanks in advanced
Jose
version of Asterisk?
--
Moises Silva
Software Developer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON
L3R 9T3 Canada
t. 1 905 474 1990 x 128 | e. m...@sangoma.com
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Re: [asterisk-users] STUN setting in Asterisk 1.6.X
[May 26 12:26:35] WARNING[16174]: chan_sip.c:2695 ast_sip_ouraddrfor: stun failed No matter which STUN server I point to I get those messages. Am I missing some other setting? Hey Carlos, That just means the stun request failed, there are several reasons for that, I won't even try to guess. So, first try this on the Asterisk CLI: stun set debug on That should give you (and us) more information to troubleshoot why the stun request failed (also enable debug and verbosity as usual). -- Moises Silva Software Developer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. m...@sangoma.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] Calling party category
Hi, in MFC-R2 signaling there is a value Calling party category signal (e.g., normal subscriber, high-priority subscriber, operator, coin-operated telephone) How can I get that information in my Asterisk?? That depends on which MFC-R2 solution are you using for Asterisk. The 2 most known are Unicall and OpenR2. Both expose with a different Asterisk variable the category. Unicall does it through UC_CATEGORY and OpenR2 thru MFCR2_CATEGORY. These categories are usually used to detect collect calls, no much testing has been done for other categories, if you have an specific issue with either Unicall or OpenR2, write to the asterisk-r2 mailing list (http://lists.digium.com/mailman/listinfo/asterisk-r2). Is there any similar value in SIP? If there is any I am not aware of it. - Moy ___ -- 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] Sangoma Wanpipe Driver Compile for DAHDI Failure
On Sun, May 3, 2009 at 6:08 AM, Doug Lytle supp...@drdos.info wrote: Atlanticnynex wrote: 3.4.1 was the first I tried. Gives me the same error. Then I'd suggest that you file a bug report with Sangoma. Doug This is probably a compilation error due to changes in DAHDI headers, since DAHDI 2.2 is still a release candidate and NOT a final release, the Wanpipe drivers were not tested with it. Why don't you just use the latest DAHDI release (and not the release candidate)? -- Moises Silva Software Developer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. m...@sangoma.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] Sangoma Wanpipe Driver Compile for DAHDI Failure
Most likely you screwed up and just compiled dahdi release but you still have loaded the rc dahdi kernel modules. cat /sys/module/dahdi/version That will tell you for sure which dahdi module version is loaded. -- Moises Silva Software Developer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. m...@sangoma.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] Feature request: manager show events
Hi, To further improve Asterisk documentation, would approve manager show events and manager show event foo commands to be added to CLI ? Today, it is possible to list available manager commands but not to list available events, AFAIK. Regards The problem is that currently, manager events are not registered, any module is free to launch events and there is no enforcement for the events to have a clearly defined structure. Work has been done lately in trying to make the naming of headers and order to be standard, but there is not programming interface enforcing that behaviour. The available events can be extracted using grep manager_event in the asterisk source code, but I agree it would be nice to see more structure there. Moy ___ -- 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] async agi question
);
if ((res 0) || (res == AST_PBX_KEEPALIVE)) {
free_agi_cmd(cmd);
break;
In order to discard any version issues, I installed a new one from scratch
and then applied the async-agi patch only, getting the same results. By the
way, I was also able to install an asterisk 1.6.0.9 with the same
configuration and dial plan like the 1.4.18 one and it worked fine.
I hope this can be useful.
Regards
Jose
-- Moises Silva wrote :
I really think you did not recompile and reinstall after applying the
new patch. I don't see any code path where the message
[Apr 13 12:03:57] DEBUG[2755]: res_agi.c:464 launch_asyncagi: No frame
read on channel SIP/501-08279028, going out ...
Is displayed but then
ast_log(LOG_DEBUG, launch_asyncagi returned (0x%X) for chan %s\n,
returnstatus, chan-name);
is NOT displayed. In fact, there is no way you can get out of
launch_asyncagi without displaying that message. I tested this with
1.4.18 version exactly.
The fact that works for some people and not for others may be due to
different asterisk versions and/or dial plan specific issues.
Please make sure the patch was correctly applied, once that is done we
can try some other things.
--
This message was sent on behalf of cyr2...@gmail.com at openSubscriber.com
http://www.opensubscriber.com/message/asterisk-users@lists.digium.com/11929418.html
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Re: [asterisk-users] async agi question
On Mon, Apr 13, 2009 at 6:59 AM, cyr2...@gmail.com wrote: Hi Moy, thanks a lot for your fix, but I'm afraid it doesn't work. I looked your patch over and I realize the code never passes by neither of the two lines you added with returnstatus = AGI_RESULT_HANGUP. Even, it seems the execution doesn't pass by res_agi.c at all, or at least, it doesn't pass over any ast_log(LOG_DEBUG,... lines like the ones your last patch has above the returnstatus fix. Could be the execution is flowing down by an if - else - break without an ast_log(LOG_DEBUG,... line? In that case, would the returnstatus = AGI_RESULT_HANGUP be added to any places more? Below is the output log for the redirect while playing a file. As you can see, there isn't any res_agi.c output on it: [Apr 13 11:20:09] DEBUG[5804]: manager.c:2108 process_message: Manager received command 'Redirect' [Apr 13 11:20:09] DEBUG[5804]: channel.c:1378 ast_softhangup_nolock: Soft-Hanging up channel 'SIP/501-08287a00' [Apr 13 11:20:09] DEBUG[5815]: channel.c:1793 ast_settimeout: Scheduling timer at 0 sample intervals [Apr 13 11:20:09] DEBUG[5815]: pbx.c:2448 __ast_pbx_run: Extension 801, priority 0 returned normally even though call was hung up [Apr 13 11:20:09] DEBUG[5815]: channel.c:1378 ast_softhangup_nolock: Soft-Hanging up channel 'SIP/501-08287a00' [Apr 13 11:20:09] DEBUG[5815]: channel.c:1477 ast_hangup: Hanging up channel 'SIP/501-08287a00' [Apr 13 11:20:09] DEBUG[5815]: chan_sip.c:3485 sip_hangup: Hangup call SIP/501-08287a00, SIP callid 2dbe6797392cde921fb7db0b16e81...@10.0.5.20) However, if the redirect is done without playing a file, the execution does pass by res_agi.c: [Apr 13 12:03:57] DEBUG[2688]: manager.c:2108 process_message: Manager received command 'Redirect' [Apr 13 12:03:57] DEBUG[2688]: channel.c:1378 ast_softhangup_nolock: Soft-Hanging up channel 'SIP/501-08279028' [Apr 13 12:03:57] DEBUG[2755]: res_agi.c:464 launch_asyncagi: No frame read on channel SIP/501-08279028, going out ... [Apr 13 12:03:57] DEBUG[2755]: pbx.c:2427 __ast_pbx_run: Spawn extension (sip_sercom,500,0) exited non-zero on 'SIP/501-08279028' [Apr 13 12:03:57] == Spawn extension (sip_sercom, 500, 0) exited non-zero on 'SIP/501-08279028' By the way, there's another thing puzzling me: Due you said this AsyncAGI patch was done for asterisk 1.6 and not for asterisk 1.4, and Henrik Westerbeg said it had worked for it as well, (please see: http://lists.digium.com/pipermail/asterisk-users/2008-December/223009.html) then I looked over the last releases at http://bugs.digium.com/bug_view_advanced_page.php?bug_id=11282 for that AsyncAGI patch and I was able to see neither of them have the returnstatus = AGI_RESULT_HANGUP either, however, ¡they work! (as Henrik said). As you can see, I'm a bit confusing about this subject. I would thank you If you can give any guidelines about it in order to be able to investigate deeper and move forward. Thank you very much for your help Jose M Arias I really think you did not recompile and reinstall after applying the new patch. I don't see any code path where the message [Apr 13 12:03:57] DEBUG[2755]: res_agi.c:464 launch_asyncagi: No frame read on channel SIP/501-08279028, going out ... Is displayed but then ast_log(LOG_DEBUG, launch_asyncagi returned (0x%X) for chan %s\n, returnstatus, chan-name); is NOT displayed. In fact, there is no way you can get out of launch_asyncagi without displaying that message. I tested this with 1.4.18 version exactly. The fact that works for some people and not for others may be due to different asterisk versions and/or dial plan specific issues. Please make sure the patch was correctly applied, once that is done we can try some other things. ___ -- 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] wanpipe 3.2.7.1 Compiling error
Which Wanpipe version did you download? 2009/4/13 Giovanni Andrés Nopal Pascual giova...@voip.unam.mx: Hi everybody! I'm triying to install a Sangoma A200-R FXO card on a Debian Linux 5 (Lenny), 2.6.26 kernel. To install wanpipe driver I type: WANPIPE_FOLDER# ./Setup install Everything seems to be ok. There are no broken dependencies and the hardware is well detected, even zaptel is recompiled with no errors, but at the time to compile wanpipe I'm getting this error message: -- include/linux/wanrouter.h:344: error: expected specifier-qualifier-list before ‘get_info_t’ -- Could anyone help me on this. Thanks in advance. -- ._._._._._._._._._._._._._._._._._._._._ D.G.S.C.A U.N.A.M Dirección de Telecomunicaciones Proyectos Especiales e Innovación Tecnológica Giovanni Andrés Nopal Pascual www.voip.unam.mx ___ -- 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 -- I do not agree with what you have to say, but I’ll defend to the death your right to say it. Voltaire ___ -- 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] wanpipe 3.2.7.1 Compiling error
Duh, read the subject. I suggest to try 3.3.16 beta, given that is probably a kernel version issue. On Mon, Apr 13, 2009 at 7:15 PM, Moises Silva moises.si...@gmail.com wrote: Which Wanpipe version did you download? 2009/4/13 Giovanni Andrés Nopal Pascual giova...@voip.unam.mx: Hi everybody! I'm triying to install a Sangoma A200-R FXO card on a Debian Linux 5 (Lenny), 2.6.26 kernel. To install wanpipe driver I type: WANPIPE_FOLDER# ./Setup install Everything seems to be ok. There are no broken dependencies and the hardware is well detected, even zaptel is recompiled with no errors, but at the time to compile wanpipe I'm getting this error message: -- include/linux/wanrouter.h:344: error: expected specifier-qualifier-list before ‘get_info_t’ -- Could anyone help me on this. Thanks in advance. -- ._._._._._._._._._._._._._._._._._._._._ D.G.S.C.A U.N.A.M Dirección de Telecomunicaciones Proyectos Especiales e Innovación Tecnológica Giovanni Andrés Nopal Pascual www.voip.unam.mx ___ -- 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 -- I do not agree with what you have to say, but I’ll defend to the death your right to say it. Voltaire -- I do not agree with what you have to say, but I’ll defend to the death your right to say it. Voltaire ___ -- 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] Multi Frequency Cycle Timeout - E1-R2 METROTEL COLOMBIA
Glad to hear it worked for you. I'd certainly like to take a look this Monday and see why openr2 did not work for you. Moy On Fri, Apr 10, 2009 at 10:42 PM, Giovanny Magallanes gmagalla...@gmail.com wrote: Hi Moises and Steve, I tried with all protocol variants for Openr2 (AR, BR, CN, CZ, CO, EC, ITU, MX, PH, VE) and setting mfcr2_skip_category=yes, but the problem persists. I tried with Unicall and, in this way, I could make and receive calls without problems, using protocol variant BR or CO (I did not try with another variants). Moises, if you wish the next Monday (4/13) we can chat (off-list) for this issue. Thanks for your attention. Giovanny Magallanes ___ -- 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] Multi Frequency Cycle Timeout - E1-R2 METROTEL COLOMBIA
The OpenR2 CO variant was implemented using the specification found in a CAS protocols reference manual by Natural MicroSystems. It was tested by someone in Colombia but I don't remember the telco though. Unicall uses the same A-6 signal to request calling party category, which is the one openr2 is using too. It seems the telco does not like this signal and is just ignoring it and openr2 eventually times out waiting for the category. Giovanny, if the problem persist after my recommendation contact me off-list to arrange a debug session. Moy On Thu, Apr 9, 2009 at 8:02 AM, Steve Underwood ste...@coppice.org wrote: Hi, There are at least 2 R2 protocol variants in Colombia - one used by land lines, and one used by the cellular networks. Unicall implements both, and I think both have been used successfully by people in Colombia (I seem to remember debugging with people there long ago). Which is the protocol called CO in openr2? Steve. Moises Silva wrote: Sounds like a protocol variant issue. Is the telco supposed to send you ANI? You have 2 options, the first option is to try with the ITU variant, if that does not work, set the option mfcr2_skip_category=yes and see if that helps. Moy On Wed, Apr 8, 2009 at 6:06 PM, Giovanny Magallanes gmagalla...@gmail.com wrote: Hi, I have installed Elastix 1.5.2 (Barranquilla, Colombia (TELCO: METROTEL)) with a TE220P (2xE1) and TDM2400P (16FXS), openr2 is included in 1.5.2 version. The outcoming calls are ok, but with incoming call i have an error: ERROR[14972] chan_dahdi.c: Chan 2 - Protocol error. Reason = Multi Frequency Cycle Timeout, R2 State = Seize ACK Transmitted, MF state = Category Request Transmitted, MF Group = Backward Group A, CAS = 0x00 DNIS = 310, ANI = , MF = 0x20 I tried with all protocol variants availables, because seems thats the cause, but I still have the problem. elastix*CLI mfcr2 show variant Variant Code Country AR Argentina BR Brazil CN China CZ Czech Republic CO Colombia EC Ecuador ITU International Telecommunication Union MX Mexico PH Philippines VE Venezuela elastix*CLI The following link has the content of files: chan_dahdi.conf, system.conf, and a tail of /var/log/asterisk/full http://pastebin.com/f3424b319 Is this really a variant protocol problem? Any suggest? Regards, GM ___ -- 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 -- I do not agree with what you have to say, but I’ll defend to the death your right to say it. Voltaire ___ -- 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] Multi Frequency Cycle Timeout - E1-R2 METROTEL COLOMBIA
Sounds like a protocol variant issue. Is the telco supposed to send you ANI? You have 2 options, the first option is to try with the ITU variant, if that does not work, set the option mfcr2_skip_category=yes and see if that helps. Moy On Wed, Apr 8, 2009 at 6:06 PM, Giovanny Magallanes gmagalla...@gmail.com wrote: Hi, I have installed Elastix 1.5.2 (Barranquilla, Colombia (TELCO: METROTEL)) with a TE220P (2xE1) and TDM2400P (16FXS), openr2 is included in 1.5.2 version. The outcoming calls are ok, but with incoming call i have an error: ERROR[14972] chan_dahdi.c: Chan 2 - Protocol error. Reason = Multi Frequency Cycle Timeout, R2 State = Seize ACK Transmitted, MF state = Category Request Transmitted, MF Group = Backward Group A, CAS = 0x00 DNIS = 310, ANI = , MF = 0x20 I tried with all protocol variants availables, because seems thats the cause, but I still have the problem. elastix*CLI mfcr2 show variant Variant Code Country AR Argentina BR Brazil CN China CZ Czech Republic CO Colombia EC Ecuador ITU International Telecommunication Union MX Mexico PH Philippines VE Venezuela elastix*CLI The following link has the content of files: chan_dahdi.conf, system.conf, and a tail of /var/log/asterisk/full http://pastebin.com/f3424b319 Is this really a variant protocol problem? Any suggest? Regards, GM ___ -- 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 -- I do not agree with what you have to say, but I’ll defend to the death your right to say it. Voltaire ___ -- 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] async agi question
Released means no patching needed, it means it was tested and put into Asterisk tree. So, I published a patch for 1.4 so it could be used in 1.4 however the feature per se was just released for Asterisk 1.6. Moy On Tue, Apr 7, 2009 at 10:01 AM, cyr2...@gmail.com wrote: Moy, I apologize if you felt under some pressure. It wasn't my mind. I only wanted to know if either there was a mistake in my configuration, or I was failing in the procedure, or it was a bug, as you said, in order to move forward. By the way, there's a thing I don't understand: In your blog at http://www.moythreads.com/wordpress/2007/12/24/asterisk-asynchronous-agi/ , talking about AsyncAGI, there's an entry where you say: If you want the patch back-ported to 1.4, I have one here: http://www.moythreads.com/asterisk-1.4.15-async-agi.patch; However, in an earlier answer to this thread, you said: Async AGI was never released for Asterisk 1.4.X, so probably the patch you used has a bug or something ... Does it mean you are talking about different things? Can you clarify, please? Many thanks Jose Arias -- This message was sent on behalf of cyr2...@gmail.com at openSubscriber.com http://www.opensubscriber.com/message/asterisk-users@lists.digium.com/11864934.html ___ -- 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 -- I do not agree with what you have to say, but I’ll defend to the death your right to say it. Voltaire ___ -- 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] async agi question
It's a bug in the Async AGI feature. I have created a new patch http://www.moythreads.com/asterisk-1.4.18-async-agi.patch Please test it and let me know if it works for you, Moy On Tue, Apr 7, 2009 at 11:50 AM, Moises Silva moises.si...@gmail.com wrote: Released means no patching needed, it means it was tested and put into Asterisk tree. So, I published a patch for 1.4 so it could be used in 1.4 however the feature per se was just released for Asterisk 1.6. Moy On Tue, Apr 7, 2009 at 10:01 AM, cyr2...@gmail.com wrote: Moy, I apologize if you felt under some pressure. It wasn't my mind. I only wanted to know if either there was a mistake in my configuration, or I was failing in the procedure, or it was a bug, as you said, in order to move forward. By the way, there's a thing I don't understand: In your blog at http://www.moythreads.com/wordpress/2007/12/24/asterisk-asynchronous-agi/ , talking about AsyncAGI, there's an entry where you say: If you want the patch back-ported to 1.4, I have one here: http://www.moythreads.com/asterisk-1.4.15-async-agi.patch; However, in an earlier answer to this thread, you said: Async AGI was never released for Asterisk 1.4.X, so probably the patch you used has a bug or something ... Does it mean you are talking about different things? Can you clarify, please? Many thanks Jose Arias -- This message was sent on behalf of cyr2...@gmail.com at openSubscriber.com http://www.opensubscriber.com/message/asterisk-users@lists.digium.com/11864934.html ___ -- 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 -- I do not agree with what you have to say, but I’ll defend to the death your right to say it. Voltaire -- I do not agree with what you have to say, but I’ll defend to the death your right to say it. Voltaire ___ -- 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] async agi question
You have to understand that this mailing list is not free instant
support. Even more, you are using an unsupported Asterisk feature for
1.4. I will check it when I have some spare time to try to reproduce
and fix it. If you are too much in a hurry you can always contact me
off-list for paid support for this feature.
Moy
On Mon, Apr 6, 2009 at 3:15 AM, Jose Arias cyr2...@gmail.com wrote:
Hi,
I was asked for the patch and I sent it. Does anybody have any news about
this subject?
I'm willing to try a fix for 1.4 but I'd need any guidelines to do it.
Thanks in advanced
Jose
2009/4/2 Moises Silva moises.si...@gmail.com
Async AGI was never released for Asterisk 1.4.X, so probably the patch
you used has a bug or something, do you still have the patch around?
Moy
On Thu, Apr 2, 2009 at 5:44 AM, cyr2...@gmail.com wrote:
Hi Henrik,
I would like to do the same thing you are doing here. I want to
implement an external queue functionality so I need to stop a play file
launched previously with an async agi command on caller's channel, sending
the call to agent's extension.
I'm redirecting caller's channel with REDIRECT while playing is taking
place but I'm always getting a hang up on caller's channel.
I'm using:
asterisk-1.4.18
asterisk-addons-1.4.7
async agi patch 2007-12-11 10:34:12 (the one back-ported to 1.4)
Both caller and agent are using 501 and 500 extensions and the async agi
loop is waiting on 800, for example. The caller is dialing 800 where a play
file is commanded through and async agi stream file command by the
application.
The relevant part of extensions.conf follows:
exten = _5.,1,Noop(SIP call on 'sip_sercom' a ${EXTEN});
exten = _5.,n,Wait(1);
exten = _5.,n,Dial(SIP/${EXTEN},${TIMEOUTDIAL},Tto);
exten = _5.,n,Hangup();
exten = _8.,1,Noop(every thing starting 8 ${EXTEN});
exten = _8.,n,AGI(agi:async);
exten = _8.,n,Hangup();
And the redirect command the application is sending to is:
Action: Redirect
Channel: SIP/501-081f0730
Exten: 500
Context: sip_sercom
Priority: 1
Therefore, Henrik, could you show me your related dial plan and the
redirect command you are sending? I wasn't able to see what I'm getting
wrong.
thanks in advanced
Jose M Arias
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Re: [asterisk-users] Unichan wtih Te201p alarms
Use dahdi_tool to see that. On Fri, Apr 3, 2009 at 9:24 AM, criptos crip...@aullox.com wrote: I'm using a Te201p card, with unichan, I want to know if my channels are ready or in alarm... but uc show channel o uc show channels, doesn't show me anything... Any Ideas? thanks. ___ -- 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 -- I do not agree with what you have to say, but I’ll defend to the death your right to say it. Voltaire ___ -- 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] async agi question
Async AGI was never released for Asterisk 1.4.X, so probably the patch
you used has a bug or something, do you still have the patch around?
Moy
On Thu, Apr 2, 2009 at 5:44 AM, cyr2...@gmail.com wrote:
Hi Henrik,
I would like to do the same thing you are doing here. I want to implement an
external queue functionality so I need to stop a play file launched
previously with an async agi command on caller's channel, sending the call to
agent's extension.
I'm redirecting caller's channel with REDIRECT while playing is taking place
but I'm always getting a hang up on caller's channel.
I'm using:
asterisk-1.4.18
asterisk-addons-1.4.7
async agi patch 2007-12-11 10:34:12 (the one back-ported to 1.4)
Both caller and agent are using 501 and 500 extensions and the async agi loop
is waiting on 800, for example. The caller is dialing 800 where a play file
is commanded through and async agi stream file command by the application.
The relevant part of extensions.conf follows:
exten = _5.,1,Noop(SIP call on 'sip_sercom' a ${EXTEN});
exten = _5.,n,Wait(1);
exten = _5.,n,Dial(SIP/${EXTEN},${TIMEOUTDIAL},Tto);
exten = _5.,n,Hangup();
exten = _8.,1,Noop(every thing starting 8 ${EXTEN});
exten = _8.,n,AGI(agi:async);
exten = _8.,n,Hangup();
And the redirect command the application is sending to is:
Action: Redirect
Channel: SIP/501-081f0730
Exten: 500
Context: sip_sercom
Priority: 1
Therefore, Henrik, could you show me your related dial plan and the redirect
command you are sending? I wasn't able to see what I'm getting wrong.
thanks in advanced
Jose M Arias
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Re: [asterisk-users] The Redirect hangups the call while playing a file
Hello, Which Asterisk version are you using? I was unable to reproduce your problem with Asterisk 1.6.0.3, also please post details about your dial plan extensions. Moy On Mon, Mar 30, 2009 at 7:13 AM, Jose Arias cyr2...@gmail.com wrote: Hi, I'm bringing this discussion here from http://www.moythreads.com/wordpress/2007/12/24/asterisk-asynchronous-agi/ about how to manage stopping a playback on a extension previously launched with AsyncAGI and redirecting the call to another exension. If I make the Redirect without a playback, the Redirect works: http://docs.google.com/Doc?id=ahfnfrcrh3rr_30f7fzq4hd But if I make the Redirect while a playback, the Redirect fails disconnecting the call: http://docs.google.com/Doc?id=ahfnfrcrh3rr_31ghh84bkd Regards ___ -- 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 -- I do not agree with what you have to say, but I’ll defend to the death your right to say it. Voltaire ___ -- 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] codec_dahdi and Asterisk 1.6.0.6
1) How can I use codec_dahdi? Would it be useful when passing a call from one dahdi channel to another dahdi channel? It is used whenever you need G729 or G723 transcoding (or any other format supported by the Digium transcoding board). If you don't have a Digium transcoding board then you don't need that module and you can disable it in modules.conf ERROR[18854]: codec_dahdi.c:399 find_transcoders: Failed to open /dev/dahdi/transcode: No such file or directory That simply means you don't have loaded the dahdi_transcode module. If you wanted to enable it you can with modprobe dahdi_transcode, that will create /dev/dahdi/transcode device, however if you don't have a Digium transcoding board the device is useless. Is this because I do not have a hardware trancoding device? Can I safely ignore this error or is it a bug? Yes, you can ignore it, or better yet, disable dahdi_codec module in modules.conf Moisés Silva -- I do not agree with what you have to say, but I’ll defend to the death your right to say it. Voltaire ___ -- 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] R2
Is in the process of being merged. http://bugs.digium.com/view.php?id=12509 http://reviewboard.digium.com/r/40/ http://www.libopenr2.org/ Moisés Silva On Thu, Jan 15, 2009 at 9:44 AM, David fire ddf...@gmail.com wrote: hi i am reading about new codecs and new stuff to be added to asterisk. (and i say thanks to all the guys who are working to add all the new features). will be R2 added to the main core of asterisk like ISDN? Thanks David -- (\__/) (='.'=)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 -- I do not agree with what you have to say, but I'll defend to the death your right to say it. Voltaire ___ -- 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] R2
That's Digium's folks decision. It was said they wanted it for 1.6.3, but, that's not for sure, as I said, they will decide. On Thu, Jan 15, 2009 at 11:54 AM, David fire ddf...@gmail.com wrote: thanks for the answer. any idea in wich version it will be merged? thanks 2009/1/15 Moises Silva moises.si...@gmail.com Is in the process of being merged. http://bugs.digium.com/view.php?id=12509 http://reviewboard.digium.com/r/40/ http://www.libopenr2.org/ Moisés Silva On Thu, Jan 15, 2009 at 9:44 AM, David fire ddf...@gmail.com wrote: hi i am reading about new codecs and new stuff to be added to asterisk. (and i say thanks to all the guys who are working to add all the new features). will be R2 added to the main core of asterisk like ISDN? Thanks David -- (\__/) (='.'=)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 -- I do not agree with what you have to say, but I'll defend to the death your right to say it. Voltaire ___ -- 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 -- I do not agree with what you have to say, but I'll defend to the death your right to say it. Voltaire ___ -- 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] async agi question
Hello Henrik, I have not used Asterisk from a user perspective lately, but, when I added the async agi functionality, I used to control this using a manager redirect action to the same priority where the channel calls async agi, that will work like a break that re-enters the async agi loop . This, of course, requires you to save the state of the channel somehow in your program to remember that the next time that channel calls async agi the sound was already played and such. http://www.voip-info.org/wiki/index.php?page=Asterisk+Manager+API+Action+Redirect Let me know if that does not work for you and we can probably write something in res_agi.c Moy On Fri, Dec 5, 2008 at 3:01 AM, Henrik Westerberg [EMAIL PROTECTED] wrote: Hi, I am developing asterisk support for our application using the Async AGI and Asterisk-Java. One thing I haven't been able to implement is how to stop playing a sound. Something similar to StopIO for Dialogic GlobalCall or DivaStopSending for Eicon. Is there any way to achieve this today which I have missed? Or could someone give me hints on how I could implement this in the res_agi.c The command asyncagi break does stop ongoing playing but also breaks the async agi control. I only want the first. Thanks in advance, /Henrik ___ -- 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 -- I do not agree with what you have to say, but I'll defend to the death your right to say it. Voltaire ___ -- 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] Trixbox 2.6.1.13 OpenR2
There are probably other OpenR2 users that can help you in asterisk-r2 mailing list (http://lists.digium.com/pipermail/asterisk-r2/) I have not tried Trixbox, but the first release of OpenR2 will be next week, probably that will help to have Trixbox a proper support. Moy On Sat, Nov 29, 2008 at 12:12 AM, Peter Lindquist [EMAIL PROTECTED] wrote: On Sat, Nov 29, 2008 at 10:18 AM, Yuri [EMAIL PROTECTED] wrote: Good morning! I verified that the trixbox version Trixbox 2.6.1.13 has support for OpenR2, I verified in the repository that has to libraries of the project openR2, but I don't manage to do to work in the trixbox, when I type the command (it colors show channeltypes)ele no demostra the support to MFC+R2, they could help finding out which package is necessary of the trixbox and which the necessary configurations that should make! I have been installing the trixbox version 2.6.1.13 and a Digium 110p, they put in the trixbox only get to do to work in ISDN! Thank you very much Hi Yuri, I also read that 2.6.1.13 would have OpenR2 support built in but found that this was not entirely true. The library package is in the repository, but support for OpenR2 is not in the provided Asterisk package. I ended up downloading the source and recompiling from the OpenR2 site. //Peter ___ -- 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 -- I do not agree with what you have to say, but I'll defend to the death your right to say it. Voltaire ___ -- 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] OPENR2 in Thailand
Hello Peter, You can ask this better in the asterisk-r2 mailing list. I don't know of anyone that has used OpenR2 in Thailand, but I am interested in adding support for that variant. Contact me at this same e-mail address or via Google talk (my e-mail address works for MSN as well ) to discuss further details. Moisés Silva On Mon, Oct 20, 2008 at 12:58 PM, Peter Lindquist [EMAIL PROTECTED] wrote: Dear All, I'm looking for someone who has implemented OpenR2 in Thailand successfully. Any settings, advice, caveats etc. are welcome. Best regards, Peter Lindqvist www.voxion.net ___ -- 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 -- I do not agree with what you have to say, but I'll defend to the death your right to say it. Voltaire ___ -- 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] DTMF issues...
Hey Carlos, What is the best method to debug DTMF issues? Do I have to sniff the SIP packets? The best method to debug DTMF issues depend on how you receive those DTMF digits. Assuming you can use SIP INFO for the DTMF, that means the DTMF digits are not really DTMF :-), that is, is not audio, with SIP INFO the digits will be received by Asterisk as part of the SIP signaling protocol and therefore you can easily spot them using sip debug peer myxpeer. If I were you I'd try that and see what I am really receiving before drawing any other conclusion. Moy -- I do not agree with what you have to say, but I'll defend to the death your right to say it. Voltaire ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help with MFC/R2
Luis, you can join asterisk-r2 mailing list in the same way you joined asterisk-users, just go to http://lists.digium.com/ and select asterisk-r2 mailing list, there you just need to provide your e-mail address. Moy On Sat, Sep 20, 2008 at 11:48 AM, Luis Morales [EMAIL PROTECTED] wrote: Moy, How i can do to join asterisk-r2 list ? My congratulations about your article in digium blog http://blogs.digium.com/page/2/ I will collaborate in your project and give support from Venezuela. Regards, Luis Morales On Sat, Sep 20, 2008 at 7:47 PM, Moises Silva [EMAIL PROTECTED] wrote: Dae, If you can assist to my session may be we can discuss this issue you are having. I am about to add Colombia support for OpenR2, and even if you want to stick with Unicall I'd like to see what's going on there :-) Guys, just for your information, as of today, there is now an asterisk-r2 mailing list, where you may want to move this thread to (or not) :-) More and more I focus on Unicall and OpenR2 issues only, which I and other R2 users can monitor in asterisk-r2 more easily. Moy On Fri, Sep 19, 2008 at 1:10 PM, Dae Yeung Um [EMAIL PROTECTED] wrote: Yes I can see the channels with UC show channels, and it's says Idle -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Luis Morales Sent: Friday, September 19, 2008 12:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Help with MFC/R2 Not really the unicall setup must be idem. So you can see the unicall channels ? It's moises are busy i can give you support too Regards, Luis Morales On Sat, Sep 20, 2008 at 12:15 PM, Dae Yeung Um [EMAIL PROTECTED] wrote: Hi Luis, But this E1 has 30 channels, all for both directions... I must differentiate this?? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Luis Morales Sent: Friday, September 19, 2008 8:06 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Help with MFC/R2 Hi dae, Your zapata.conf must be ok, now inyour unicall.conf [channels] language=es context=from-pstn usecallerid=yes hidecallerid=no immediate=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=yes rxgain=0.0 txgain=0.0 loglevel=255 callgroup=1 pickupgroup=1 group=1 musiconhold=default relaxdtmf=yes category=NATIONAL_SUBSCRIBER ; protocolclass=mfcr2 protocolvariant=co,20,4,x,T1=1500,T2=24000,T3=15000,max-seize-wait-ack=3000 protocolend=cpe ; ; E1 IN group = 1 context = from-pstn channel = 1-15 ; E1 OUT group = 2 context = from-pstn channel = 17-31 - It's very important that you identify: - E1 lines in - E1 lines out Now from your cosole type: asterisk -r now from asterisk cli type: pbx* UC show channels The result must be an list with your unicall channels, similar to: Channel Extension Context Status Language MusicOnHold 1from-pstn Idle es default 2from-pstn Idle es default 3from-pstn Idle es default 4from-pstn Idle es default 5from-pstn Idle es default 6from-pstn Idle es default 7from-pstn Idle es default 8from-pstn Idle es default 9from-pstn Idle es default 10from-pstn Idle es default 11from-pstn Idle es default 12from-pstn Idle es default 13from-pstn Idle es default 14from-pstn Idle es default 15from-pstn Idle es default 17 6842 from-pstn Idle es default 18from-pstn Idle es default 19from-pstn Idle es default 20from-pstn Idle es default 21from-pstn Idle es default 22from-pstn Idle es default 23from-pstn Idle es default 24from-pstn Idle es default 25from-pstn Idle es default 26from-pstn Idle es default 27from-pstn Idle es default 28from-pstn Idle es default 29from-pstn Idle es default 30from-pstn
Re: [asterisk-users] Help with MFC/R2
, no module. # Global data loadzone= us defaultzone = us [Channels] language=en usecallerid=yes echocancel=yes rxgain=0 txgain=0 group=1 callgroup=0 pickupgroup=0 amaflags=default accountcode=avantel musiconhold=default context=from-pstn group=1 loglevel=0 protocolclass=mfcr2 protocolvariant=ar,20,4 channel = 1-15 channel = 16-31 I cannot receive calls... I cant see any type of logs on the console when I try to call in. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Luis Morales Sent: Thursday, September 18, 2008 12:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Help with MFC/R2 I'm not sure but on E1 setup you can have only one way (in or out). In my case i have 15 in and 15 out. Told me more about your hardware: - E1 cards - How did you do to connect E1 interface to E1 asterisk's card ? - You can receive calls ? Please send us zapata.conf and unicall.conf Regards, Luis Morales On Fri, Sep 19, 2008 at 12:46 PM, Dae Yeung Um [EMAIL PROTECTED] wrote: All channels 1~15, 17~31 is supposed to be double way. To place and receive calls. The line is supposed be E1-MFC/R2 and works perfect with a Panasonic PBX, actually Exists any variant of MFC/R2? And how can I configure it to get working? Your help will be very appreciated! Thank you! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Luis Morales Sent: Thursday, September 18, 2008 10:10 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Help with MFC/R2 Ok, in your E1 setup: 1-15: to outgoing calls 16-30: for incomming calls ? Now for make calls your telephone company must be provide MFC-R2 signaling. In your case the logs files show an invalid signal on make call. Regards, Luis Morales On Fri, Sep 19, 2008 at 10:10 AM, Dae Yeung Um [EMAIL PROTECTED] wrote: Hello I got: [Sep 17 19:24:50] DEBUG[4934] chan_unicall.c: unicall_call called - 'g1/6055151' [Sep 17 19:24:50] DEBUG[4934] chan_unicall.c: unicall_call caller id - '1102' [Sep 17 19:24:50] DEBUG[4934] chan_unicall.c: no UC_CATEGORY specified for chan UniCall/1-1, using default NATIONAL_SUBSCRIBER [Sep 17 19:24:50] VERBOSE[4934] logger.c: MFC/R2 UniCall/1 Call control(1) [Sep 17 19:24:50] VERBOSE[4934] logger.c: MFC/R2 UniCall/1 Make call [Sep 17 19:24:50] WARNING[4934] chan_unicall.c: Make call failed - Blocked [Sep 17 19:24:50] VERBOSE[4934] logger.c: -- Couldn't call g1/6055151 [Sep 17 19:24:50] VERBOSE[4934] logger.c: MFC/R2 UniCall/1 Channel gains [Sep 17 19:24:50] VERBOSE[4934] logger.c: MFC/R2 UniCall/1 Channel switching [Sep 17 19:24:50] DEBUG[4934] chan_unicall.c: Hangup: channel: 1 index = 0, normal = 11, callwait = -1, thirdcall = -1 [Sep 17 19:24:50] DEBUG[4934] chan_unicall.c: Updated conferencing on 1, with 0 conference users [Sep 17 19:24:50] VERBOSE[4934] logger.c: -- Hungup 'UniCall/1-1' -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Luis Morales Sent: Thursday, September 18, 2008 8:17 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Help with MFC/R2 Dae, Activate debug full: asterisk -vr in other console do: tail -vf /var/log/asterisk/full Try to put call and send us more details about your logs Regards, Luis Morales On Thu, Sep 18, 2008 at 11:49 PM, Dae Yeung Um [EMAIL PROTECTED] wrote: In fact I see 1101 in the rx bits on all channels... But I have in parallel one old Panasonic Key Phone system (Actually in production, to be replaced by asterisk), and it's works perfectly and immediately once I pass the E1 cables to there... So, the problem is not from Telco... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Moises Silva Sent: Wednesday, September 17, 2008 10:57 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Help with MFC/R2 It seems to me your lines are blocked. Execute zttool and if you see 1101 in the rx bits, it means the telco (or whatever you have in the other end) has blocked their side. If this is a telco line you need to call them and tell them to unblock your lines. On Wed, Sep 17, 2008 at 10:33 PM, Dae Yeung Um [EMAIL PROTECTED] wrote: Thank you for the reply I shutdown asterisk and tried again and I have to following logs... OUTGOING TEST : Testcall.conf caller yes destination-no 6055151 originating-no 7309130 protocol-class mfcr2 protocol-variant ar,20,4 circuits 1-2 Log: ./testcall Chan 1, class 'mfcr2', variant 'ar,20,4', end 2, caller 1, from '7309130' to '6055151' Chan 2, class 'mfcr2', variant 'ar,20,4', end 2, caller 1, from
Re: [asterisk-users] Help with MFC/R2
Do as Luis says, however, I feel that as long you keep getting 1101 Unicall won't work. AFAIK The only IDLE bit pattern recognized by libmfcr2 as IDLE is 10XX, as long you have 11 in the first 2 bits (AB), libmfcr2 will report the lines as blocked. On Thu, Sep 18, 2008 at 8:16 AM, Luis Morales [EMAIL PROTECTED] wrote: Dae, Activate debug full: asterisk -vr in other console do: tail -vf /var/log/asterisk/full Try to put call and send us more details about your logs Regards, Luis Morales On Thu, Sep 18, 2008 at 11:49 PM, Dae Yeung Um [EMAIL PROTECTED] wrote: In fact I see 1101 in the rx bits on all channels... But I have in parallel one old Panasonic Key Phone system (Actually in production, to be replaced by asterisk), and it's works perfectly and immediately once I pass the E1 cables to there... So, the problem is not from Telco... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Moises Silva Sent: Wednesday, September 17, 2008 10:57 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Help with MFC/R2 It seems to me your lines are blocked. Execute zttool and if you see 1101 in the rx bits, it means the telco (or whatever you have in the other end) has blocked their side. If this is a telco line you need to call them and tell them to unblock your lines. On Wed, Sep 17, 2008 at 10:33 PM, Dae Yeung Um [EMAIL PROTECTED] wrote: Thank you for the reply I shutdown asterisk and tried again and I have to following logs... OUTGOING TEST : Testcall.conf caller yes destination-no 6055151 originating-no 7309130 protocol-class mfcr2 protocol-variant ar,20,4 circuits 1-2 Log: ./testcall Chan 1, class 'mfcr2', variant 'ar,20,4', end 2, caller 1, from '7309130' to '6055151' Chan 2, class 'mfcr2', variant 'ar,20,4', end 2, caller 1, from '7309131' to '6055152' Loading protocol mfcr2 Thread for channel 0 Thread for channel 1 MFC/R2 Chan 1: Call control(9) MFC/R2 Chan 1: Unblock MFC/R2 Chan 1: 1001 - [1/BLOCKED /Idle /Idle ] MFC/R2 Chan 2: Call control(9) MFC/R2 Chan 2: Unblock MFC/R2 Chan 2: 1001 - [1/BLOCKED /Idle /Idle ] MFC/R2 Chan 1: local_unblocking_expired Chan 1: -- Local end unblocked! :-) Chan 1: -- Local end unblocked! :-) MFC/R2 Chan 2: local_unblocking_expired Chan 2: -- Local end unblocked! :-) Chan 2: -- Local end unblocked! :-) Main thread Incoming test : Testcall.conf caller no protocol-class mfcr2 protocol-variant ar,20,4 on-offered answer circuits 1-2 Log: Chan 1, class 'mfcr2', variant 'ar,20,4', end 2, caller 0, from '' to '' Chan 2, class 'mfcr2', variant 'ar,20,4', end 2, caller 0, from '' to '' Loading protocol mfcr2 Thread for channel 0 Thread for channel 1 MFC/R2 Chan 1: Call control(9) MFC/R2 Chan 1: Unblock MFC/R2 Chan 1: 1001 - [1/BLOCKED /Idle /Idle ] MFC/R2 Chan 2: Call control(9) MFC/R2 Chan 2: Unblock MFC/R2 Chan 2: 1001 - [1/BLOCKED /Idle /Idle ] MFC/R2 Chan 2: local_unblocking_expired MFC/R2 Chan 1: local_unblocking_expired Chan 2: -- Local end unblocked! :-) Chan 1: -- Local end unblocked! :-) Chan 1: -- Local end unblocked! :-) Chan 2: -- Local end unblocked! :-) Main thread Main thread Main thread Seems no any response from far side... Do you have any ideas?? Only one time, I got the following log: #./testcall Chan 1, class 'mfcr2', variant 'ar,20,4', end 2, caller 0, from '' to '' Chan 2, class 'mfcr2', variant 'ar,20,4', end 2, caller 0, from '' to '' Loading protocol mfcr2 Thread for channel 0 Thread for channel 1 MFC/R2 Chan 1: Call control(9) MFC/R2 Chan 1: Unblock MFC/R2 Chan 1: 1001 - [1/BLOCKED /Idle /Idle ] MFC/R2 Chan 2: Call control(9) MFC/R2 Chan 2: Unblock MFC/R2 Chan 2: 1001 - [1/BLOCKED /Idle /Idle ] Chan 2: -- Protocol failure on channel 0, cause (32773) Unexpected CAS bit pattern Chan 1: -- Protocol failure on channel 0, cause (32773) Unexpected CAS bit pattern Chan 2: -- Protocol failure on channel 0, cause (32773) Unexpected CAS bit pattern Chan 1: -- Protocol failure on channel 0, cause (32773) Unexpected CAS bit pattern MFC/R2 Chan 2: local_unblocking_expired MFC/R2 Chan 1: local_unblocking_expired Chan 1: -- Local end unblocked! :-) Chan 1: -- Local end unblocked! :-) Chan 2: -- Local end unblocked! :-) Chan 2: -- Local end unblocked! :-) Main thread MFC/R2 Chan 2: - 1101 [1/BLOCKED /Idle /Idle ] Chan 2: -- Far end blocked! :-( Chan 2: -- Far end blocked! :-( MFC/R2 Chan 1: - 1101 [1/BLOCKED /Idle /Idle ] Chan 1: -- Far end blocked! :-( Chan 1: -- Far
Re: [asterisk-users] Help with MFC/R2
The line is supposed be E1-MFC/R2 and works perfect with a Panasonic PBX, actually Exists any variant of MFC/R2? And how can I configure it to get working? As I said, no matter which variant you try, the AB bits MUST be in 10 to be able to make calls with Unicall/libmfcr2. I have never seen a variant which does not set bits AB in 10 for IDLE and 11 for BLOCKED. Most variants differ in the MF tones used, not on the R2 bits. Which telco is this and which country? you used Argentina, are you there? I am willing to troubleshoot your box if you give me access. You can contact me at google talk or msn at the same address you see in this e-mail. Moy -- I do not agree with what you have to say, but I'll defend to the death your right to say it. Voltaire ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help with MFC/R2
That means someone else has already open the zap device, most likely Asterisk. Just one application at a given time can open a zap device. You cannot run testcall and Asterisk at the same time unless you make sure they don't try to open the same channels. Moy On Wed, Sep 17, 2008 at 1:27 AM, Dae Yeung Um [EMAIL PROTECTED] wrote: Hello I'm new in this list, but I have some experience working with asterisk and we are located in Bogota, Colombia. At now I'm having some problems configuring an E1 MFC/R2. I have configured Unicall as indicated using the following versions, and everything seems to be correct, but I can't place or receive the calls (receive CHANUNAVAIL message) asterisk-1.4.21.2 libsupertone-0.0.2 spandsp-0.0.5 chan-unicall-1.4-r01 libunicall-0.0.6 libmfcr2-0.0.6 zaptel-1.4.11 Inside asterisk, I can see the all UC channels active and Idle: asterisk*CLI UC show channels Channel Extension Context Status Language MusicOnHold 9from-pstn Idle en default 10from-pstn Idle en default 11from-pstn Idle en default (omitted) 34from-pstn Idle en default 35from-pstn Idle en default 36from-pstn Idle en default 37from-pstn Idle en default 38from-pstn Idle en default 39from-pstn Idle en default I configured the testcall.conf to check: destination-no 6055151 protocol-class mfcr2 protocol-variant ar,20,4 protocol-end cpe on-offered accept circuits 9-10 But the system answer with: # ./testcall Chan 9, class 'mfcr2', variant 'ar,20,4', end 1, caller 0, from '' to '6055151' Chan 10, class 'mfcr2', variant 'ar,20,4', end 1, caller 0, from '' to '6055152' Loading protocol mfcr2 Failed to open channel: Device or resource busy Anyone has any Idea why I can't place or receive a call? Your help will be really appreciated! DAE YEUNG UM ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- I do not agree with what you have to say, but I'll defend to the death your right to say it. Voltaire ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help with MFC/R2
It seems to me your lines are blocked. Execute zttool and if you see 1101 in the rx bits, it means the telco (or whatever you have in the other end) has blocked their side. If this is a telco line you need to call them and tell them to unblock your lines. On Wed, Sep 17, 2008 at 10:33 PM, Dae Yeung Um [EMAIL PROTECTED] wrote: Thank you for the reply I shutdown asterisk and tried again and I have to following logs... OUTGOING TEST : Testcall.conf caller yes destination-no 6055151 originating-no 7309130 protocol-class mfcr2 protocol-variant ar,20,4 circuits 1-2 Log: ./testcall Chan 1, class 'mfcr2', variant 'ar,20,4', end 2, caller 1, from '7309130' to '6055151' Chan 2, class 'mfcr2', variant 'ar,20,4', end 2, caller 1, from '7309131' to '6055152' Loading protocol mfcr2 Thread for channel 0 Thread for channel 1 MFC/R2 Chan 1: Call control(9) MFC/R2 Chan 1: Unblock MFC/R2 Chan 1: 1001 - [1/BLOCKED /Idle /Idle ] MFC/R2 Chan 2: Call control(9) MFC/R2 Chan 2: Unblock MFC/R2 Chan 2: 1001 - [1/BLOCKED /Idle /Idle ] MFC/R2 Chan 1: local_unblocking_expired Chan 1: -- Local end unblocked! :-) Chan 1: -- Local end unblocked! :-) MFC/R2 Chan 2: local_unblocking_expired Chan 2: -- Local end unblocked! :-) Chan 2: -- Local end unblocked! :-) Main thread Incoming test : Testcall.conf caller no protocol-class mfcr2 protocol-variant ar,20,4 on-offered answer circuits 1-2 Log: Chan 1, class 'mfcr2', variant 'ar,20,4', end 2, caller 0, from '' to '' Chan 2, class 'mfcr2', variant 'ar,20,4', end 2, caller 0, from '' to '' Loading protocol mfcr2 Thread for channel 0 Thread for channel 1 MFC/R2 Chan 1: Call control(9) MFC/R2 Chan 1: Unblock MFC/R2 Chan 1: 1001 - [1/BLOCKED /Idle /Idle ] MFC/R2 Chan 2: Call control(9) MFC/R2 Chan 2: Unblock MFC/R2 Chan 2: 1001 - [1/BLOCKED /Idle /Idle ] MFC/R2 Chan 2: local_unblocking_expired MFC/R2 Chan 1: local_unblocking_expired Chan 2: -- Local end unblocked! :-) Chan 1: -- Local end unblocked! :-) Chan 1: -- Local end unblocked! :-) Chan 2: -- Local end unblocked! :-) Main thread Main thread Main thread Seems no any response from far side... Do you have any ideas?? Only one time, I got the following log: #./testcall Chan 1, class 'mfcr2', variant 'ar,20,4', end 2, caller 0, from '' to '' Chan 2, class 'mfcr2', variant 'ar,20,4', end 2, caller 0, from '' to '' Loading protocol mfcr2 Thread for channel 0 Thread for channel 1 MFC/R2 Chan 1: Call control(9) MFC/R2 Chan 1: Unblock MFC/R2 Chan 1: 1001 - [1/BLOCKED /Idle /Idle ] MFC/R2 Chan 2: Call control(9) MFC/R2 Chan 2: Unblock MFC/R2 Chan 2: 1001 - [1/BLOCKED /Idle /Idle ] Chan 2: -- Protocol failure on channel 0, cause (32773) Unexpected CAS bit pattern Chan 1: -- Protocol failure on channel 0, cause (32773) Unexpected CAS bit pattern Chan 2: -- Protocol failure on channel 0, cause (32773) Unexpected CAS bit pattern Chan 1: -- Protocol failure on channel 0, cause (32773) Unexpected CAS bit pattern MFC/R2 Chan 2: local_unblocking_expired MFC/R2 Chan 1: local_unblocking_expired Chan 1: -- Local end unblocked! :-) Chan 1: -- Local end unblocked! :-) Chan 2: -- Local end unblocked! :-) Chan 2: -- Local end unblocked! :-) Main thread MFC/R2 Chan 2: - 1101 [1/BLOCKED /Idle /Idle ] Chan 2: -- Far end blocked! :-( Chan 2: -- Far end blocked! :-( MFC/R2 Chan 1: - 1101 [1/BLOCKED /Idle /Idle ] Chan 1: -- Far end blocked! :-( Chan 1: -- Far end blocked! :-( Main thread Main thread Main thread But after rerunning the test, I only get the first log (w/o Far end replies.) Any help will be really appreciated! Thank you! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Moises Silva Sent: Wednesday, September 17, 2008 8:33 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Help with MFC/R2 That means someone else has already open the zap device, most likely Asterisk. Just one application at a given time can open a zap device. You cannot run testcall and Asterisk at the same time unless you make sure they don't try to open the same channels. Moy On Wed, Sep 17, 2008 at 1:27 AM, Dae Yeung Um [EMAIL PROTECTED] wrote: Hello I'm new in this list, but I have some experience working with asterisk and we are located in Bogota, Colombia. At now I'm having some problems configuring an E1 MFC/R2. I have configured Unicall as indicated using the following versions, and everything seems to be correct, but I can't place or receive the calls (receive CHANUNAVAIL message
Re: [asterisk-users] Help-ASTERISK-MFCR2
You need to enable loglevel=255 in unicall.conf and enable all the levels of logging in logger.conf, otherwise the logs you post don't say much. Moisés Silva On Fri, Jun 6, 2008 at 2:58 PM, Mariano Borgognone [EMAIL PROTECTED] wrote: Dears, I have problem ASTERISK with PSTN SIEMENS EWSD (MFC R2), I don´t receive call for PSTN, i don´t understand why. please i need your help # MFC/R2 normalmente no usa CRC4 span=1,1,0,cas,hdb3 cas=1-15:1101 dchan=16 cas=17-31:1101 loadzone=us defaultzone=us [channels] usecallerid=yes hidecallerid=no callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no musiconhold=default protocolclass=mfcr2 protocolvariant=ar,10,10 protocolend=cpe group = 1 context= e1-incoming channel = 1-15 channel = 17-31 ;skip time slot 16 Here is the LOGS when I try do make calls Jun 6 16:02:18 WARNING[5060]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 - 0001 [1/ 1/Idle /Idle ] Jun 6 16:02:18 WARNING[5060]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Detected Jun 6 16:02:18 WARNING[5060]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Making a new call with CRN 32769 Jun 6 16:02:18 WARNING[5060]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 1101 - [2/ 2/Idle /Idle ] Jun 6 16:02:18 WARNING[5060]: chan_unicall.c:2644 handle_uc_event: Unicall/1 event Detected Jun 6 16:02:30 WARNING[5060]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 - 1001 [2/ 2/Seize ack /Seize ack] Jun 6 16:02:30 WARNING[5060]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Far end disconnected(cause=Normal, unspecified cause [31]) - state 0x2 Jun 6 16:02:30 WARNING[5060]: chan_unicall.c:2644 handle_uc_event: Unicall/1 event Far end disconnected Jun 6 16:02:30 WARNING[5060]: chan_unicall.c:2930 handle_uc_event: CRN 32769 - far disconnected cause=Normal, unspecified cause [31] Jun 6 16:02:30 WARNING[5060]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Call control(6) Jun 6 16:02:30 WARNING[5060]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Drop call(cause=Normal Clearing [16]) Jun 6 16:02:30 WARNING[5060]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Call disconnected(cause=Normal, unspecified cause [31]) - state 0x800 Jun 6 16:02:30 WARNING[5060]: chan_unicall.c:2644 handle_uc_event: Unicall/1 event Drop call Jun 6 16:02:30 WARNING[5060]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Call control(7) Jun 6 16:02:30 WARNING[5060]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Release call Jun 6 16:02:30 WARNING[5060]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 1001 - [1/1000/Clear fwd /Seize ack] Jun 6 16:02:30 WARNING[5060]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Release guard expired Jun 6 16:02:30 WARNING[5060]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Destroying call with CRN 32769 Jun 6 16:02:30 WARNING[5060]: chan_unicall.c:2644 handle_uc_event: Unicall/1 event Release call -- Unicall/1 released Jun 6 16:02:30 WARNING[5060]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Channel echo cancel Best Regards, Mariano Borgognone ___ -- 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 -- I do not agree with what you have to say, but I'll defend to the death your right to say it. Voltaire ___ -- 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] Unicall - How to automatically block collect calls
The latest version of the driver included in http://www.moythreads.com/astunicall/ comes with a change that will set the variable UC_CATEGORY in your dialplan, Brasil has a special category for those calls, don't remember the name that will show up, but you can make a couple of tests and then drop any call with that specific category. Moy On Mon, May 5, 2008 at 9:14 AM, Oscar Patricio [EMAIL PROTECTED] wrote: Hi! I am using asterisk with unicall in brasil. Everything was working fine, but now we want to set up a way to automatically drop collect calls, because we have an IVR answering all calls automatically! Can you tell me, what I have to configure to block collect calls in the asterisk? Thank you! Best regards, Óscar Patrício ___ -- 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 -- I do not agree with what you have to say, but I'll defend to the death your right to say it. Voltaire ___ -- 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] MFC/R2 in chan_zap , Testers Wanted
If you are an MFC/R2 user and want to help in the development of chan_zap support for this signalling, please take a look at the bugtracker at http://bugs.digium.com/view.php?id=12509 and/or contact me. Currently just México support is built-in, if you want your country variant supported, drop me a line. Moisés Silva -- I do not agree with what you have to say, but I'll defend to the death your right to say it. Voltaire ___ -- 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] MFC/R2 in chan_zap , Testers Wanted
Hello Moisés, thanks for your effort on this! I would love to use Digium cards for MFC/R2 signalling in the future. Currently you can use Digium cards with Unicall :-) , tho, having MFC/R2 on chan_zap is more handy. I added some info you might like in the bugtracker, you might take a look at it. I will, thanks! Moisés Silva -- I do not agree with what you have to say, but I'll defend to the death your right to say it. Voltaire ___ -- 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] MFC/R2 in chan_zap , Testers Wanted
Hello Ruben, Yes, if you consider using R2 support in chan_zap Unicall is no longer required. I will be not available online this weekend, please let me know your feedback after your try it. We can also meet via MSN so I can assist you in testing the next weeked (3-4 May). Thanks for the help. Moisés Silva On Thu, Apr 24, 2008 at 8:26 AM, Ruben Zamora [EMAIL PROTECTED] wrote: Moises Thats means, that we arent going to use unicall? If that true i can test these weekend with a E1-Axtel. Thanks Ruben Moises Silva escribió: If you are an MFC/R2 user and want to help in the development of chan_zap support for this signalling, please take a look at the bugtracker at http://bugs.digium.com/view.php?id=12509 and/or contact me. Currently just México support is built-in, if you want your country variant supported, drop me a line. Moisés Silva ___ -- 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 -- I do not agree with what you have to say, but I'll defend to the death your right to say it. Voltaire ___ -- 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] MFC/R2 in chan_zap , Testers Wanted
Way more handy and will be much more reliable too. Steve Underwood did a great job implemeting it, but as far as I know the code isn't actively maintained anymore. Of course your implementation of MFC/R2 will take a while to become stable, but hey -- it's a start. Agreed. Russel pointed some licensing stuff related to the Digivoice drivers. Please listen to him on that, I had no idea of that kind of complication. If your implementation of MFC/R2 can't be integrated in Zaptel, then it's no much better than Unicall. You mean chan_zap/zapata (zaptel is the kernel code). I had already discussed with Russell the licensing. He just got confused for a second because he forgot LGPL code requires both license files (GPL and LGPL). I guess if you look at Digivoice's code to figure out how it works and then write your own code, there will be no licensing issues. But that's just a guess, Russel will need to clarify it. I did not discussed that with Russell, but I will. In the meantime, since I am aware of the licensing concerns I have not even looked at that code :-) I would like to test the BR variant the next week, I will contact you off-list to see if we can meet via IM. - Moisés Silva -- I do not agree with what you have to say, but I'll defend to the death your right to say it. Voltaire ___ -- 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] MFC/R2 in chan_zap , Testers Wanted
Unicall MFC/R2 is activelly maintained. by Moy. Actually it's a backport of the Steve driver (now coded for Callweaver derivative) to Asterisk (1.2, 1.4, and 1.6 soon). It works pretty well. In fact, it works more stable in 1.4 than the original Steve driver in 1.2, and with better sound under heavy loads. The Asutunicall page can be found here: http://www.moythreads.com/astunicall/ Hum, wonder who this moy is hey wait, that's me! . Even when is in my plans to keep giving general maintenance to chan_unicall, my long term plan is to leave R2 support into chan_zap, so I would recommend to all users to try chan_zap R2 support, the more users we get the faster the driver will be stable enough to replace chan_unicall, the less headaches you will have (I hope). - Moy or Moisés Silva, same shit :-) -- I do not agree with what you have to say, but I'll defend to the death your right to say it. Voltaire ___ -- 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] Problems with Quality Voice in a Asterisk-E1-Unicall
The E1 use ALAW, if you want to avoid trans-coding use ALAW in your phones as well. In any call you have 2 call legs, callee and caller, try to isolate the problem and determine if the audio is really coming that bad from the E1, you can use ztmonitor to hook into the E1 and listen to the audio. If the audio you get using ztmonitor is deficient then you know it has nothing to do with trans-coding or the codec you use in your phones. Is the Digium card missing interrupts? (zttool will tell you so) Moisés Silva On Sun, Apr 20, 2008 at 3:23 PM, Ruben Zamora [EMAIL PROTECTED] wrote: I Have with Asterisk- Unicall - E1 (MFC/R2). Days before a install a Digium Card TE122P with hardware echo cancelation, these because a had a echo in some in and out calls. I replaced the card. I no more echo but in my conversation the voice start to doing things. Like after a minutes i start hearing the voice cut. or the cant hearme.. I remove in the zaptel.conf the echotraining.I dont know if i really need to do these changes in the unicall.conf.??? In my Asterisk am using GXP2020 Grandstream what is better ulaw,alaw,g729??? I apreciate any help. Thanks Ruben ___ -- 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 -- I do not agree with what you have to say, but I'll defend to the death your right to say it. Voltaire ___ -- 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] G729 license count...
http://store.digium.com/productview.php?product_code=G729CODEC http://www.digium.com/en/docs/G729/g729policy.php http://www.voip-info.org/wiki-Asterisk+G.729+Licensing On Thu, Apr 17, 2008 at 11:14 AM, Carlos Chavez [EMAIL PROTECTED] wrote: I need a refresher course on how many licenses I need to buy. I have an Asterisk server that receives calls by SIP (G729) and then sends them to the PSTN via 32 Zap interfaces on an Astribank. I cannot remember if the license is per channel or per call so I do not know if I need 32 or 64 licenses for this application. Could anyone please remind me? -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- I do not agree with what you have to say, but I'll defend to the death your right to say it. Voltaire ___ -- 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] Unicall + incomplete DNIS on international calls
Hello Ivan, I don't see nothing wrong in terms of signaling. When your side (Asterisk/Unicall) request ANI, the other end answer with the signal F, which means No More ANI, hence you receive an empty ANI string. When your side request DNIS, the other end does not answer in several seconds, which means 'No More DNIS' then your side timesout and resume the signaling to proceed to Answer if appropriate. You should definitely setup a call with Telmex support and see what do they see on their side. No timer tweaking will help for the ANI here because it is pretty clear they are sending End of ANI signal 'F'. In the case of DNIS I also think the timer will not help, you already waited 15 seconds to timeout for DNIS, if they don't send DNIS in 15 seconds is pretty obvious they are not going to send any. Moy On Tue, Apr 1, 2008 at 4:40 PM, Iván Reyes Tejera [EMAIL PROTECTED] wrote: Hello everybody, i'm from Mexico, at the time i´m working on a production server with asterisk 1.2.25 + spandsp-0.0.4 + libmfcr2-0.0.3+libsupertone-0.0.2+libunicall-0.0.3 and zaptel-1.2.22. I installed this version of astunicall that i downloaded from http://www.moythreads.com/astunicall/ Everything works fine, i'm able to make outgoing calls and recive incoming calls with all ANI and DNIS digits, except for International incoming call. My phone provider(Telmex) gives me 10 digits of ANI and 4 digits of DNIS, that i´ve configured on my unicall.conf. My main issue becomes when i recive an internationall incoming call, there is no ANI, appears with only one digit instead four, and that digit it's always a number 1( i attach unicall log). I already talked with my phone provider about this issue, and, as they told me, all DNIS and ANI of international incoming calls are just bypassed by them directly to my server. They mentioned something about timers that may avoid my server to recive all values (DNIS and ANI), but i'm not quite sure about this. On my file unicall.conf i added some timers that moises commented on his forum. Any clue what would be the reason of my issue ? Here are my files: -- unicall.conf -- [channels] language=en context=from-pstn usecallerid=yes hidecallerid=no callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no echotraining=800 relaxdtmf=no rxgain=0 txgain=0 group=1 callgroup=0 pickupgroup=0 immediate=no callerid=asreceived amaflags=default musiconhold=default protocolclass=mfcr2 protocolvariant=mx,10,4,7,t1=15000,t2=24000,t3=15000,max-seize-wait-ack=2000 channel=1-10 loglevel=255 -- zapata.conf --- [channels] context=default usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=no echocancelwhenbridged=no echotraining=no relaxdtmf=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no --- zaptel.conf --- loadzone=us defaultzone=us #Sangoma A101 port 1 [slot:0 bus:10 span:1] wanpipe1 span=1,1,0,cas,hdb3 cas=1-10:1101 dchan=16 - DEBUG UNICALL --- Mar 31 13:10:35 DEBUG[14902] chan_unicall.c: MFC/R2 UniCall/8 - 0001 [1/IDLE/Idle /Idle ] Mar 31 13:10:35 DEBUG[14902] chan_unicall.c: MFC/R2 UniCall/8 Detected Mar 31 13:10:35 DEBUG[14902] chan_unicall.c: MFC/R2 UniCall/8 Creating a new call with CRN 32769 Mar 31 13:10:35 DEBUG[14902] chan_unicall.c: MFC/R2 UniCall/8 1101 - [2/DETECTED/Seize ack /Seize ack] Mar 31 13:10:35 NOTICE[14902] chan_unicall.c: Unicall/8 event Detected Mar 31 13:10:35 DEBUG[14902] chan_unicall.c: MFC/R2 UniCall/8 - 1 on [2/DETECTED/Seize ack /Seize ack] Mar 31 13:10:35 DEBUG[14902] chan_unicall.c: MFC/R2 UniCall/8 6 on - [2/DETECTED/Group C /Category req ] Mar 31 13:10:35 DEBUG[14902] chan_unicall.c: MFC/R2 UniCall/8 - 1 off [2/DETECTED/Group C /Category req ] Mar 31 13:10:35 DEBUG[14902] chan_unicall.c: MFC/R2 UniCall/8 6 off - [2/DETECTED/Group C /Category req ] Mar 31 13:10:35 DEBUG[14902] chan_unicall.c: MFC/R2 UniCall/8 - 2 on [2/DETECTED/Group C /Category req ] Mar 31 13:10:35 DEBUG[14902] chan_unicall.c: MFC/R2 UniCall/8 1 on - [2/DETECTED/Group C /ANI request ] Mar 31 13:10:35 DEBUG[14902] chan_unicall.c: MFC/R2 UniCall/8 - 2 off [2/DETECTED/Group C /ANI request ] Mar 31 13:10:35 DEBUG[14902] chan_unicall.c: MFC/R2 UniCall/8 1 off - [2/DETECTED/Group C /ANI request ] Mar 31 13:10:35 DEBUG[14902] chan_unicall.c: MFC/R2 UniCall/8 - F on [2/DETECTED/Group C /ANI request ] Mar 31
Re: [asterisk-users] DTMFR2- UNICALL
Well, then it is irrelevant what version of unicall, spandsp, asterisk
etc you are using.
All that it could matter is your zaptel version and zaptel.conf
configuration. I ignore why you need to control the bit pattern at
that level, but AFAIK zaptel only let you control the CAS ABCD bits in
E1 timeslot 16 using ioctl()'s. Regarding bits in timeslot 0 and MFAS,
I suppose that is in complete control of the zaptel drivers. You have
the code available to dig into it.
- Moisés Silva
On Fri, Mar 7, 2008 at 8:52 AM, Jessica Gonzalez Arriagada
[EMAIL PROTECTED] wrote:
The standard G.732 :
Multiframe Aligment Signal:
When Timeslot 16 of the E1 frame is used for Channel Associated Signaling
purposes, Frame 0 contains information that is used by the receiver to
identify the incoming frame. Specifically, this pattern in Timeslot 0,
Frame 0 is called the Multiframe Alignment Signal (MFAS).
| Bit 1 | Bit 2 | Bit 3 | Bit 4 | Bit 5 | Bit 6 | Bit 7 |
Bit 8 |
Frame 0 | --- | --- | --- | --- | --- | --- |
--- | |
|0 |0 |0 |0| X| Y| X
|X|
X = Spare Bits, set to 1 if not used.
Y = Yellow Alarm (Loss of MultiFrame Alignment Signal)
(0 = Normal | 1 = Loss of MFAS)
But, I used a analyser and I saw :
and must be 1000
i need to change the bit X
Regards,
Jessi
On 3/6/08, Moises Silva [EMAIL PROTECTED] wrote:
What kind of problems are you talking about and what you want to modify?
On Thu, Mar 6, 2008 at 2:42 PM, Jessica Gonzalez Arriagada
[EMAIL PROTECTED] wrote:
Hi Asterisk-user, Steve;
I´m using libmfcr2-0.0.3.tar.gz,
libsupertone-0.0.2.tar.gz,libunicall-0.0.3.tar.gz,spandsp-0.0.3pre22.tgz
with Fedora core 6 ,Asterisk 1.2.14, libpri 1.2.4 , zaptel 1.2.20; So
everything is working perfectly with MFCR2, but sometimes i have
problems
with MultiFrame Alignment Signal (MFAS),i´m using standard G.732..I
would
like to Know, where can i mofify this??
Regards,
Jessi
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Re: [asterisk-users] DTMFR2- UNICALL
What kind of problems are you talking about and what you want to modify? On Thu, Mar 6, 2008 at 2:42 PM, Jessica Gonzalez Arriagada [EMAIL PROTECTED] wrote: Hi Asterisk-user, Steve; I´m using libmfcr2-0.0.3.tar.gz, libsupertone-0.0.2.tar.gz,libunicall-0.0.3.tar.gz,spandsp-0.0.3pre22.tgz with Fedora core 6 ,Asterisk 1.2.14, libpri 1.2.4 , zaptel 1.2.20; So everything is working perfectly with MFCR2, but sometimes i have problems with MultiFrame Alignment Signal (MFAS),i´m using standard G.732..I would like to Know, where can i mofify this?? Regards, Jessi ___ -- 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 -- I do not agree with what you have to say, but I'll defend to the death your right to say it. Voltaire ___ -- 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
