similar to: Dropping duplicate answer

Hi all, We are running Asterisk 1.0.0 with a TE410P. Very often we exerience calls dropping in the middle of the call. I enable the full logging and saw a couple of suspicious messages right before the hangup. Thos could happen on a Zap-IAX2 bridge as well as on a Zap-Agent bridge... I see Nov 23 09:08:36 DEBUG[-1274020944]: Bridge stops because we're zombie or need a soft hangup:

Maybe of you guys know the answer to this: We have T1's that come from both MCI and Global Crossing as channelized (24 Ports per T) with inband (DTMF) delivery of ANI and DNIS (format = *DNIS*ANI*). My old equipment was set for D4, AMI, SF and Wink Start and so is Asterisk. I've moved these T's to Asterisk TE410P and inbound calls are arriving to external voice mail system

Maybe of you guys know the answer to this: We have T1's that come from both MCI and Global Crossing as channelized (24 Ports per T) with inband (DTMF) delivery of ANI and DNIS (format = *DNIS*ANI*). My old equipment was set for D4, AMI, SF and Wink Start and so is Asterisk. I've moved these T's to Asterisk TE410P and inbound calls are arriving to external voice mail system correctly

Maybe one of you can help me with this: We have T1's that come from both MCI and Global Crossing as uses channelized (24 Ports per T) with inband (DTMF) ANI and DNIS delivery (format = *DNIS*ANI*). My old equipment was set for D4, AMI, SF and Wink Start and so is Asterisk Server. I've moved these T's to Asterisk TE410P and inbound calls are arriving to external voice mail

I'm a bit confused about how to handle this. I have Asterisk sitting in the middle between a Qwest Long Distance T1 (Voice T1, D4, SF, AMI) and an external voice mail PC using a Dialogic D/240SC-T1 card. The Qwest T1 originally was connected to the Dialogic card directly. The signaling was set to E&M Wink Start because Dialogic used this as its default settings, so it just worked

Hi all, I am experiencing a problem with the quality of the voice communication between an IAX based softphone (WinIAX) and an outside line through a FXO port or even with a regular analog phone connected to a FXS port. The party using the IAX softphone hears his own echo a plit of a second after speaking. The party on the analog end does not experience any echo. I tried to modify the KFLAG

Hi all I'm trying to configure a TE410P in Europe with three E1s and a T1 channel bank (already bought in the US) to receive and make calls through *. I managed to get the E1s to work, but I'm having trouble with the channel bank (a Rhino). I have tried the bank in spans 1 and 4 changing jumpers in the card (always with no PRI connected, as I have no PRI where I'm testing now,

Hi, Ok, I put the new TE410 into our system last night - This is the second TE410P card. There is also TDM400P in Slot 5 with IRQ 9 which is shared with aic7xxx (SCSI) According to "cat /proc/interrupts", the new card is in the second slot using IRQ 11 (the first card is in slot 1 on IRQ 10 - but always has been). With only the original card install, there has been no issues with

Hi all, We have an asterisk server running on redhat 9, with a TE410P card connected to 4 PRIs, on a dual 2.6GHz xeon server with 2GB of RAM. We also have 10-12 agents using IAX based softphones on winxp. Almost all of the usage consists of outbound calls done automatically through the PRIs and connected to an agent once answered. The agents perform the 'ready' operation via IAX by

I have a TE410P with two real Telco T1's and the other 2 ports terminate into an in-house voice mail/IVR system. Calls arrive from Telco are routed to the appropriate in-house system based on the DID Digits. This part works perfectly. Now I what to allow the in-house VMS to dial though asterisk to the Telco T1's. The VMS complains there is no dial tone on its T1 and drops the attempt.

Hello, Sorry for a bit offtopic but it seems the bogofilter community is just as dead as dspam is. I trying to switch from DSPAM to bogofilter on a server where if I turn bogofilter on I run into the following issue: Certain users have sender and recipient bcc maps setup in postfix. This is for forwarding copy of an email to other address for example: Incoming lclude at company.com ->

Hello, Yes but how to activate this for all users, not on a per user basis or without writing out thousands of sieve files to users directories. Thank you. On 2017-06-15 13:33, Stephan Bosch wrote: > Op 6/15/2017 om 12:19 PM schreef Luescher Claude: >> I have +vnd.dovecot.duplicate plugin enabled but it seems to be >> ineffective. Any other tips for a workaround? >> >

I have echocancel=yes in zapata.conf but when I do a zap show channel 1, I notice echo cancellation is turned off. I followed the article that talks about the order in which the statements need to be in zapata.conf to get echo canceling to work: http://lists.digium.com/pipermail/asterisk-users/2005-June/110615.html But it is still not working. Does anyone know how to get echo

Hi I have kind of an odd problem. When dialing in from an outside line via a TE410P card it seems like MusicOnHold and MP3Player doesn't work properly (for me anyway). The remote end which is calling * doesn't hear the music but just keeps ringing. But if I insert a Playback("file_which_dont_exist") just before the Moh or MP3Player I can hear the music. Actually I observed the

Op 6/15/2017 om 12:19 PM schreef Luescher Claude: > I have +vnd.dovecot.duplicate plugin enabled but it seems to be > ineffective. Any other tips for a workaround? > Enabling that extension does nothing by itself. You need to add a corresponding rule to your Sieve script. Also, that is the deprecated name for that extension. It is explained fully here:

I'm stuck on 1.2 until I can pass DTMF from a SIP Trunk (Vitelity Virtual PRI) call towards a ZAP (TE410P using e&m wink) port. The call connects OK, I can hear DTMF with DNIS & ANI inband from asterisk to the external IVR, Voice is OK, but if any DTMF is required after the bridge has been made, they are muted. I posted on http://bugs.digium.com/view.php?id=12913 but I have got

Hi I would like to eliminate a large number of lines of the dataframe df1 The lines to delete are given here by the values of Mat (ex : 2,4,7,10). but I have a large number (300) values of Mat dput(df1) structure(list(Mat = c(1, 1, 1, 1, 1, 1, 2, 2, 2, 2, 3, 3, 3, 3, 3, 4, 4, 4, 4, 4, 5, 5, 5, 5, 5, 5, 6, 6, 6, 6, 6, 6, 7, 7, 7, 7, 7, 7, 7, 8, 8, 8, 8, 8, 8, 8, 9, 9, 9, 9, 9, 9, 10, 10, 10, 11,

Greetings, According to the FAQ "When you load the module and have no circuit/channel bank the LED's should flash red" I get the "knight rider" lights before the module loads, but after the modules are loaded I don't get any lights, other equipment plugged in or not. The card has been on my shelf a while so I wonder if it needs a firmware upgrade ? Can the cards

I've installed dovecot with maildir format in /home/claude/mail. I fetch my email with fetchmail and they are placed in /var/spool/mail/claude. How can I transfer them automatically in /home/claude/mail/INBOX/new How can I setup something to transfer my email automatically as they arrive. Claude

We have had quite a bit of success with our T100P and TE410P cards interfacing to Nortel Meridian PBXes and also to a Livingston Portmaster 3 using ESF/B8ZS and various combinations of E&M wink and ISDN PRI (usually in 5ESS mode). In the near future, I may also need to interface to a Definity via a T1. I was planning to use PRI--is that an option for you on your switch? -----Original