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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. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090919/082c8d8a/attachment.htm

Hello, I recently switched over to using Mixmonitor versus Monitor to see if it would clear up some warble that I was getting in my recordings. It did indeed clear that up, but a new problem was introduced. The recordings for no reason will just end abnormally. There is no rhyme or reason as to when they will end, but usually after a minute or so. Here is my current setup. Asterisk v. 1-2-0rc2

...he fax tone and jusmps to the fax extension and rxfax application starts and the max machine starts the fax but saddenly stops and seems the rxfax have died. It doesnt returns, not files in the output dir and .......... Anyone have any idea or help how could i get whats wrong here. Regards. M. Shokuie Nia _________________________________________________________________ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/

...ce my call routing mechanisms. Is SER applicable of doing that or should i write any application on the SER to do so ro is there any need for a softswitch at all? Or as a more basical question is there any need for SER, Asterisk cant do it itself? Any help and hints would be highly appreciated, M. Shokuie Nia.

Dear folks, I have a problem with console/dsp using ALSA. I dont know why the output sound is choppy sometimes and also the input one has an awful delay. Is there anyone here with experince about ALSA channels or not? I would be highly appreciated if anyone could help me. Regards. M. Shokuie Nia _________________________________________________________________ Don't just search. Find. Check out the new MSN Search! http://search.msn.click-url.com/go/onm00200636ave/direct/01/

.... Moreover when i made a tiff file using Microsoft mdi, everything works fine but on the other end of the call, the received fax is shrinked in size. Anyone has any idea whats the right file format and compression type for it? PS. Im using libtiff-3.7.1-2 and spandsp-0.0.2-pre25 Regards. --- M. Shokuie Nia _________________________________________________________________ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/

...carrier down app_rxfax.c:76 span_message: FLOW Trainability test failed - longest run of zeros was 1696 ..... chan_zap.c:4351 __zt_exception: Exception on 23, channel 1 chan_zap.c:3539 zt_handle_event: Got event On hook(1) on channel 1 (index 0) app_rxfax.c:329 rxfax_exec: Got hangup Regards. M. Shokuie Nia _________________________________________________________________ Don't just search. Find. Check out the new MSN Search! http://search.msn.click-url.com/go/onm00200636ave/direct/01/

Did you ever get an answer to this problem ? I too am seeing this and it's driving me mad !!! Jim -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061019/85af372e/attachment.htm

Hello, I recently purchased a ArtDio IPF-2600 phone from voipsupply.com, but they did not include a manual. Does anyone know the default login information? I have tried all of the common ones that I can think of. If anyone knows, it would be greatly appreciated. Thanks! _________________________________________________________________ Stay in touch with old friends and meet new ones with

...aming and HDB3 line coding but dont know which signalling to use for channels. I'd use 3 bit CAS signalling and 20 incoming channels and 10 outgoing ones. Anyone could help me define the signalling for these channels. PS. Im using Sangoma cards. Any help would be highly appreciated. --- M. Shokuie Nia. _________________________________________________________________ Don't just search. Find. Check out the new MSN Search! http://search.msn.click-url.com/go/onm00200636ave/direct/01/

...yone send me a working sample of wanpipe.conf and zaptel.conf for cas signalling? and is it possible that the alarms are because of looping the links (although in ccs mode it works just fine) ? Any help and hint would be highly appreciated. PS. i define the span in zaptel as cas with hdb3 -- M. Shokuie Nia. _________________________________________________________________ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/

...ad it manually or restart it, but it would work fine afterward, no problem how many times you stop and start the *. It seems, there is a missequence of deamon loading at boot time but i have no clue which deamons! Im using FC5, MySQL5, Asterisk 1.2.18 Any help would be highly appreciated. --- M. Shokuie Nia. _________________________________________________________________ Connect to the next generation of MSN Messenger? http://imagine-msn.com/messenger/launch80/default.aspx?locale=en-us&source=wlmailtagline

...ound choppy and I'm hearing hers the same but not all the time during a call, sound sometimes are clear. Even when I'm putting the sip side on hold i hear the same choppy music on hold. Any one have any idea how i could get closer to the problem. Any hint would be highly appreciated. -- M. Shokuie Nia. _________________________________________________________________ Discover the new Windows Vista http://search.msn.com/results.aspx?q=windows+vista&mkt=en-US&form=QBRE

Dear Folks, Im looking for a way to detect if an analog line is connected to card or not (Im using Sangoma A200). Im using the dialtone detection when dialing but need a way to detect the disconnection of the line when it actually happens. Anyone have any hints or tricks for this? Regards. -- Mohammad Sh. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, I try configuring Quintum DX gateway as link to PSTN for *. Now, I can dial number which is connect to Quintum, and call is diverted to *. I don't know what I should set, if I want call from SIP_phone registred in Asterisk to PSTN via Quitnum. I set in sip.conf account for Quintum [sip_proxy-out] type=peer outboundproxy=QUINTUM_IP , and changed extensions.conf. When

Hi guys, This should be has an easy answer for you, my users are complaining that when they press # and then ear gorgeous Allison "Transfer" the timeout is very small, they must enter immediatly the extension to transfer the call. Is it possible to change this? ;transferdigittimeout => 3 ; Number of seconds to wait between digits when transfering a call This is timeout

Hi all! We've released VoiceOne 0.4.0, a web-based and open source solution which allows to fully manage an Asterisk service hosted on a LAMP server. We focused on an charming and overall user-friendly interface. Thanks to the authentication based on roles, once configured by a super user, the PBX may be easily maintained even by an Asterisk unskilled users. From a technical point of

Hi all, Strange request, I have a customer where we are putting an Asterisk PBX in front of a legacy (non-VoIP) PBX. One of the requirements it that the Asterisk PBX have 2 PRI ports (on towards the legacy PBX and one towards the carrier) with the ability to go to pass through should the Asterisk PBX (software or hardware level) fail. I did not see this feature in the Digium, Sangoma, Allo, or

I have been doing a lot of reading forums and elsewhere but am somehow unable to connect the dots. Here is what I am trying to accomplish initially and then wish for it to grow bigger from here on. I have two POTS (Analog) line that would connect to the Asterisk Box. I have, to begin with 5 IP phones (PoE), all connected to a switch. Asterisk Box with a LAN card also connects to the same switch.