similar to: SIP expiry time

Hi, I currently have a E100P card installed on my machine and the E1 subscription will be activated pretty soon. However, I have no idea how to configure asterisk to make inbound and outbound call using the E1. Especially for extensions.conf. Below is the configuration I used for zaptel.conf and zapata.conf. Is it possible if someone can verify if the configuration for zaptel and zapata is

Hi all, I have the following setup PSTN -> ASTERISK -> IVR (using dialogic card) 1) Caller id information is presented to asterisk during the first and second ring. 2) Hence, Asterisk waits for 2 rings before pickup the call and forwarding to the appropriate FXS port. 3) The IVR application also waits for 2 rings before picking up the call to get the caller id. 4) Hence any caller

----- Original Message ----- From: <asterisk-users-request@lists.digium.com> To: <asterisk-users@lists.digium.com> Sent: Wednesday, December 01, 2004 7:07 AM Subject: Asterisk-Users Digest, Vol 5, Issue 6 > Send Asterisk-Users mailing list submissions to > asterisk-users@lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, visit >

Hi, I have a problem with the queue cmd. I am trying to redirect an incoming call to another phone when nobody in the queue answer it within 18 seconds. Somehow the incoming call keeps on retrying within the queue. The second part was never executed. Below is a part of my extensions.conf and the queue.conf. Is there someone out there who can help me? Thanks in advance. Regards,

Hi there, I tried to get a few "Optipoint 400 SIP" working with *, but it refused to work properly. In my testing-network i have two Sjphones (they are working really fine) and three optipoints. I?m able to dial the number of a Sjphone on all of the optipoints. The call is signalled at the Sjphone with the right number of the optipoint and an connection can be established. But when I

Hi Rowland, and many thanks for fast reply, When using --noexpiry, the userAccountControl is set to 66048, which disable expiry for password as well (in MS console, "password never expires" is now checked). This means that the password expiry (let say, every 6 month) will never popup again to the user, which is in my sense a wrong behaviour. Is there a way to change ONLY

Hi list :) I am looking for the right command to achieve my goal. I would like to remove the account expiry date of an ACCOUNT with a samba-tool command (account never expires) Options of "samba-tool user setexpiry" are : --filter=FILTER LDAP Filter to set password on --days=DAYS Days to expiry --noexpiry Unfortunately, the "noexpiry" parameter just set another option

Hi, I'm getting this error when registering with SIP server using Asterisk 1.4.10 and Freepbx... I'm getting this error no matter what I try to setup in sip.conf : - I'm getting confused whether options are maxexpirey=36000 or maxexpiry=36000 ? - Can I solve this with some settings in sip.conf or is this problem harder ? - I've read something about Asterisk's bug on this

Hi all, I have a couple of Samba 4 DCs on my network and I created a new service account LDAPReader on my DCs that my non-Samba third-party services such as Redmine successfully use to access AD via the LDAPS protocol. I have a couple of questions that relate to having service account of this nature implemented in Samba and I wondered if the group could possibly provide some advice? 1)

Hi, i have configured a Samba PDC based on idealx.org. now, whenever i set the sambaMustChangePassword flag to 0, then from the subsequent logon, there is a popup urge me for changing password. now, the problem is after i have changed the password, the sambaMustChangePassword is set to 2147483647(unix timestamp), which if i converted it into human readable format, it will be 2038 year,

Hi Elliot, I couldn't find anything related to smbpasswd expiry. Since u have "unix password sync = true", just a wild guess, if u could turn off password ageing in unix passwd file (man passwd for more detail) and see if problem persists. Rgds Gary Elliot wrote: > Hi guys... I setup samba 2.2.5 as a PDC ... I have w2k clients. It seems > that now I am prompted to change

I have tried this patch (against 3.5p1) and would very much like it to be in the OpenSSH 3.6p1 release, if possible: http://bugzilla.mindrot.org/show_bug.cgi?id=14 On that note, I'd like the Sun BSM patch to be included also, if possible. I have it working applied to 3.5p1: http://bugzilla.mindrot.org/show_bug.cgi?id=125 In fact, both patches work together, apparently. If I have any

Thanks for taking a look greg and hank. This seems to be getting bettre everyday..help please My sjphone is running on the same box as asterisk...i believe then the red hat firewall should not be a problem. Whenever i dial from CLI i get ######### Executing Goto("OSS/dsp", "default|s|1") in new stack -- Goto (default,s,1) -- Executing Wait("OSS/dsp",

I am having a problem with SJphone registration, having read the list and wathced it for a while for similar problems. I just can't seem to figure out the problem. I tryed to follow a tutorial from http://www.voip-info.org/tiki-index.php?page=Asterisk+phone+sjphone, but in SJphone (SIP tab), I can't find the following setting. Use local outbound proxy - checked. Proxy IP Address:

Hi. I have sjphone installed on windows and working except for dtmf. I read the docs for sjphone and it uses inband dtmf. I configired dtmfmode=inband but it still does not recognize it. Someone on the lists said that inband only works using alaw or ulaw but i tried only allowing that too but still no go. Hmm.. any other ideas? I can't get any other client to work on windows :-/ I

Dear all expert, (i posted this question one time, but i couldn't reach the answer -so allow me to post here) 1)I download asterisk realse version 1.2 beta1. After that i compile it successfully and run it with: asterisk -vvvc 2)I follow the instruction in http://www.asteriskguru.com/tutorials/asterisk_voip_ipphone.html in sip.conf: i add two account: [ivan] type=friend username=ivan

Greetings, I'm having a job getting asterisk to register with a Huawei softX3000 softswitch via SIP. I keep getting 401 Unauthorized. Funny thing is I can successfully register SJPhone, a PA1688 IP Phone as well as a WiFi Phone against the switch without *any* problems. I think it's got to be something as simple as perhaps the register string which is currently

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configuration ISDN BRI card : ISDN Olitec PCI 128 (hisax gazel) internet connection by ISDN 64kb/s dynamic IP nom de domaine registered to : dyndns.org avec ddclient to register IP par ddclient asterisk (on internet gateway) configuration pour ISDN BRI par virtual modems /dev/ttyI* (modem.conf) logical telephone SIP "SJPHONE" on 2 local stations "windows" (i don't succeed