similar to: Retry dial when busy

This looks like a fun box... a Voip to Dect gateway, I've mailed them for pricing details.... < <http://www.computex.com.tw/news_archive_detail.asp?index=4053> http://www.computex.com.tw/news_archive_detail.asp?index=4053> Betel Consultancy Abelenlaan 19 T: +31 20 640 3018 1185 RT Amstelveen E: <mailto:info@betel.nl> info@betel.nl The Netherlands W:

Simple Q but I can't find the answer in the archives (and am too lazy to look in the source, but then its 32 Celcius here... Do all digium cards provide the zapata timing? e.g. also the analogs (including the X100P) or only the E1/T1 -ones or do I need to use ztdummy on the analog cards? Thanks, Michiel Betel Consultancy Abelenlaan 19 T: +31 20 640 3018 1185 RT Amstelveen

Hi all, I'm using asterisk CVS-08/14/03-22 on a box with a digium T1 connected to a channel bank and a digium E1 connected to the PSTN. I get occasional warnings from asterisk: WARNING[37909]: File chan_zap.c, Line 3197 (zt_read): zt_rec: Unknown error 500 This happens mosttimes in a loop like this: [netland_helpdesk] exten =>

I just downloaded the latetst CVS. A compile now complains about a missing srv.c & srv.h used in chan_sip.c. Can they be added? -- Betel Consultancy Abelenlaan 19 1185 RT Amstelveen The Netherlands http://www.betel.nl tel. +31 621 858 469

Hi all, Has anyone already written something which allows saving and loading the internal DB settings? All users CFWD and speeldial settings are stored in the DB in my setup which makes it a pain to restart Asterisk.... Looking at showtree in db.c (why isn't that exposed in the CLI?) It shouldn't be too difficult, but I don't want to reinvent the wheel. On the same track, I am also

Hi, as the current debian maintainer of the openssh package is a bit busy, I'm helping him with fixing a part of the bugs in openssh that debian users found will forward some of the reports to you. This is the first one and a fix or a comment why this should not be fixed would be appropriated. Thanks Space in filename is not correctly passed by scp to other invoked programs:

About every hour I see the yellow alarms on all or a number of channels of my PRI which is connected to the dutch telephony network, Asterisk keeps on working fine.... Here's an example where channel 1-24 went into alarm: WARNING[90124]: File chan_zap.c, Line 4139 (handle_init_event): Detected alarm on channel 1: Yellow Alarm WARNING[90124]: File chan_zap.c, Line 4139 (handle_init_event):

Dear sir Schwartz, In response to a granted online request to receive R code in order to generate Theil's Uncertainty coefficient, I was hoping I could receive the same favor. https://stat.ethz.ch/pipermail/r-help/2011-May/279210.html Thank you in advance, I hope to hear from you. Kind regards, Jos? Snoep Stagiair Universitair | MC ES - SOFY +31 6 13060740 Snoep.Jose at

Hi All, I was able to track down what I believe is a bug when using AGI services. This bug may crash your system if your extensions.conf script is intensive in using AGI services. Depending on your system's ulimit, * keeps opening files until it reaches the system limit and then stops responding. Function app_agi/launch_script seems to leave an open and unused file. Can someone confirm this?

Hello, How would I go ahead a record all phone calls into and out of my asterisk server. I know the legality issues behind it, but I could always play a recording to let people know they will be recorded. Brian J. Schrock Network Engineer, RHCE, CCNA Anistone Technologies Phone: 614-537-2817 FAX: 614-573-7165 6926 Avery Rd. Dublin, OH 43017

I have had * installed for a couple of weeks now and am very impressed. I have got Zap, SIP and MGCP working and can call freely between them with just things like transfer still to sort out etc. I then though I would add H.323 support to my working system, having read the previous threads on the subject before I installed I installed the pre-reqs pwlib openh323 gnugk for h.323 gatekeeper

Hi, We are using threewaycalling & flash transfers over a CAC channelbank. The following happens: Call comes in to my extension I talk to a party and press flash party goes on hold, I get get dail tone I dial internal number internal party answers I press flash once more we are now in a three party conference Or I hang up, and thus transfer the call. Thats fine, but.... What if the

asterisk-users-request@lists.digium.com wrote: >Send Asterisk-Users mailing list submissions to > asterisk-users@lists.digium.com > >To subscribe or unsubscribe via the World Wide Web, visit > http://lists.digium.com/mailman/listinfo/asterisk-users >or, via email, send a message with subject or body 'help' to > asterisk-users-request@lists.digium.com > >You can

Does there exist a text file with all the 'standard' Asterisk voice messages? I'm planning to get them recorded in dutch, but need to know the exact text of each prompt... Michiel

I have a number of Zap/ extensions defined in a queue with ringall strategy. When this queue is called sometimes Asterisk seems to think that one of these channels is busy, while it is NOT. The following is shown on the console: --Called 44 -- Called 36 -- Called 41 -- Called 35 -- Called 38 -- Zap/44-1 is ringing -- Zap/36-1 is ringing -- Zap/41-1 is ringing

The following transfer led to a crash of asterisk, without leaving a core or any utterances in messages or debug file. It looks like the zombie which was created during the MASQ-transfer was not cleaned up... But why did it start a Dial??? And... why does Asterisk die when this happens?? Thanks!!! Michiel -- Zap/32-1 answered Zap/6-1 -- Stopped music on hold on Zap/6-1 -- Starting

Hello, I am having some trouble with the Asterisk::AGI perl library. It seems that the AGI->Exec() command is causing me a problem. Here's the line in my AGI code: $AGI->exec('Record',"$vmfile:wav, 30"); I'm trying to record voicemail to the file name stored in $vmfile with a silence timeout of 30. However, this is not being parse by AGI or Asterisk correctly,

Does anyone have any recommendations on implementing Answering Machine detection for call generation programs? What I would like is * to determine what picks up the other line (Answering Machine, Voicemail, or Human) to determine which action to take. For example: If * detects Answering Machine or Voicemail, hangup call & the AGI will log (ANSWERING MACHINE DETECTED) and at that point,

I had a small cluster with a disperse 3 volume. 2 nodes had hardware failures and no longer boot, and I don't have replacement hardware for them (it's an old board called a PC-duino). However, I do have their intact root filesystems and the disks the bricks are on. So I need to rebuild the cluster on all new host hardware. does anyone have any suggestions on how to go about doing this?

Hi, I don't know if it's possible to stop a STREAM FILE pushing a key. Anyone know it? Best. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030507/04d18302/attachment.htm