similar to: SIP pickupgroup

Hi, I am not an this list, I hope it gets through: We just moved from dovecot-1.2.12 to dovecot-2.0.6. One of our users wants his mail directly in his homedir which worked fine so far but with 2.0.6: Nov 09 15:19:25 imap(foo.bar at example.com): Error: user foo.bar at example.com: Initialization failed: Initializing mail storage from mail_location setting failed: No home directory for system

Hi all - I have the latest samba up and running fine except for one small problem. The username I use to log into the NT domain is different than the username I log into samba with. I have no control over the former, and would *really* rather not change the latter. I've searched the net, and the archives, and got only saw bits and pieces of this issue, never quite fully answered. Is it

I have just started to play with callgroups and pickupgroups.. I updates my * from CVS this morning (about 15 mins ago).. I have placed callgroup=1 and pickupgroup=1 into each of my 3 phone configurations in sip.conf.. I place a call from phoneA to phoneB, then I go to phoneC and dial *8# , the call does not get picked up by phoneC and continues to ring on phoneB.. Have I not configured

Hi, I asked a time ago what were callgroup and pickup group used for. I have done some proofs and all, and I'm not sure if I have pick the idea up well!! That's what I understand: For example: group=1 callgroup =2 and pickupgroup=2 and my phone is a membership of the group 1. that's mean that when a phone that belong to group 2 is ringing, I'll be able to answer this call dialing

HI,ALL, I have multiple PSTN lines registered as multiple SIP channels (e.g. SIP/line1, SIP/line2, SIP/line3, etc...), on the multiple gateways( I uses the SIPURA3000). I wants to arrange them into an ordered hunting group for outbound calling. I used http://www.voip-info.org/wiki/view/hunt-dial+macro for reference. My configure files like blew.

I'm trying to set up asterisk so any phone connected to channel 1-16 of my Adit600 channel bank can pick up a call coming in on channel 24. I do not wish to ring any of the 16 channels on an incoming call -- this is strictly so I can pick up the line if I see it ringing and wish to answer at work. I have channel 24 in call group 3, and channels 1-16 in pickup groups 1 and 3. However

callgroups & pickupgroups greater than 31 are not working for sip calls with 1.2.14 tarball. Anyone know which branches support 64? John

Hi, I currently have deadtime = 15 in my smb.conf. This featured worked good to disconnect clients that have been idle for > 15 minutes in the past. With the 3.0.13 version, it seems to do nothing. As in, it just keeps the user connected indefinitely. Here is what I see when no one is currently at the office and everyone is logged off their computer. Thanks [root@spicy p]#

I have problems using the call pickup under Asterisk 1.8.4.2. I have another Asterisk with 1.6 - and it is working fine with the same settings. I have setup the same callgroup and pickupgroup for all extensions in sip.conf - just to make things simple for testing. The sequence *8 seems to be completely ignored by Asterisk - the client shows "Call answered" when dialing *8 while the

Hello group! I?m new to Asterisk but got one running finally :) Now I?m trying to solve following problem. I have company Automated Attendant and each employee have SIP phone at home, SIP phone in office, cell phone. I want all those 3 phones to be ?one?. So, if someone calls our company number and dials my extension - I?d like 3 phones to ring at the same time. What is this feature and where

Hi all. I've made a patch for chan_oss.c to enable callgroups and pickupgroups in it (since wasn't enabled). I needed it for a special use of the console (pickup calls arriving to the console from another phone) btw, If someone is interested, I can submit a patch to the bugtracker. I won't do it until that's usefult for someone... since is a very special features that probably no

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, Before I hack out the ',' -> '|' tr in extension.conf parser, any way to escape ',' that I missed? - -- Regards, Tais M. Hansen ComX Networks Tel: +45-70257474 Fax: +45-70257374 -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.2.2 (GNU/Linux) iD8DBQE/ZfuP2TEAILET3McRAjXUAJ0VjuFeABe5jqpSlrBakDC2IMjvrQCfcBYU

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, How can I check if (i.e.) my provider is requesting me to hide the callerid? I.e. (Telco)E1/PRI---Zap(Asterisk1)IAX---IAX(Asterisk2)SIP---EP Now, if a call comes from the Telco with CLI screening prohobited to Asterisk1, where the call is forwarded using Dial() via IAX to Asterisk2 and then (also using Dial()) on to a SIP endpoint, how do I

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, As I understand it, IAX2 trunking requires zaptel timing. Zaptel timing is provided by Zaptel cards, ztdummy or ztrtc. ztdummy requires usb-uhci and ztrtc can't run on smp systems. So if you only have smp systems with ohci and no zaptel cards (because it's a sip/iax2 gateway) then you're screwed? - -- Regards, Tais M. Hansen

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, Seems like Asterisk/chan_sip in some special cases gets it's rtp channels mixed up. I've got a few reports on users hearing someone elses conversation on the line. Could be port problems, but I haven't had time to make any traces or tests yet. Before I start to analyse this periodic problem, I thought I'd just check with the

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, When making a call like the one below, I get double DTMF tones on the PSTN side. DTMF tones sent from the PSTN arrives squelched on the SIP side. SIP > Asterisk2 > IAX2 > Asterisk1 > ZAP > PSTN SIP has been configured to use rfc2833 on both the SIP endpoint and the Asterisk. SIP endpoint also suggests a payload value of 101.

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I'm having a problem with the following expression examples. exten => s,1,NoOp($[$[${value} >= 10] & $[${value} < 18]]) exten => s,1,GotoIf($[$[${value} >= 10] & $[${value} < 18]]?3) ${value} is 13 in both examples above. First extension evaluates to 1 while second evaluates to 0 even though it's the same

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi. Using inAccess Networks chan_oh323, I'm experiencing some clicks or pops, how can I fix that? - -- Regards, Tais M. Hansen ComX Networks Tel: +45-70257474 Fax: +45-70257374 -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.2.2 (GNU/Linux) iD8DBQE/QMfF2TEAILET3McRAu9zAJwNWtv+QSpka0NGaVk9E/IDHyalhwCgkQME Gynfp5zF0SWZUQEjelp7sBI= =CSqT

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, As I understand it, Asterisk currently uses the timestamps in incoming RTP packets to build outgoing voice frames. Is this true? Would it be possible for me to use i.e. zaprtc as a timing source for the outgoing stream? I.e. in a setup like below I'd like to use zaprtc timing on Ast1 because I don't trust the timestamps coming from

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, Now that I've been unable to register 2 hardware SIP phones and one software (Kphone), I'm beginning to doubt that chan_sip works at all. - -- Regards, Tais M. Hansen ComX Networks Tel: +45-70257474 Fax: +45-70257374 -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.2.2 (GNU/Linux)