similar to: ATA 188 and fax

I use cisco ATA 186 ( Version: v2.16 ) with sip protocol and asterisk as pbx. I need feature called as 'three way calling' or 'transfer with consultation'. Registering,calling and 'blind transfer' work fine. Is this feature provided by sip clients or by asterisk itself ? What I have to configure in ATA and what keys I have to press on my phones ? Three way calling is

I see that there's now a 2.16.1 upgrade path for Cisco ATA-186 devices, dated (variously) July 11 or July 14 2003. Here are some interesting bugs that claim to be fixed. Most notable is CSCeb17953, at least from my perspective, as I've hit this bug before. CSCea42480 The Cisco ATA ignores the Require:100rel header and processes call. CSCea69889 The Cisco ATA builds a 302 Moved

Should I expect a standard fax machine connected to an ata-188 connected to an asterisk server, connected to a pri fed from a cisco 7206vxr to work correctly? It needs to have a standard fax machine, receiving and emailing it won't be acceptable. Thanks dave -- Dave Weis "I believe there are more instances of the abridgment djweis@sjdjweis.com of the freedom of the

Hey all! I had an experience trying to set up an E1 in Brazil which could help somebody. In Brazil is very common telcos to have just R2 digital as their primary signaling. As I were trying to set up an E100P, which does not support R2 yet, I had to test an other signaling which works perfectly with Asterisk. They call this signaling as RDSI, using ccs as framing and PA (primary access) as

Hi, Is there is some form of VAD on * for SIP channels, cause I have a problem with MOH. I made an extension which simply plays MOH, when I dial that extension with my ATA188 MOH sounds choppy if I talk on the phone the MOH keeps playing. I saw the sip channel (show channel SIP/*) and I see no packets going in/out when I talk then packets shows going in/out. I don?t have this kind of problem

Hello, I am trying to see what I need to do SIP to H323 using G.729. I have Oh323 and SIP working with G711 fine. If I have a SIP client configured to use G729 and H323 client also to G729, how many license should I need to buy from Digium. Many thanks SW

Hi everybody,, I need to block incoming collect calls to my Asterisk box but I could not find out where to do that. Went to zaptel.h but I did not see any timing which can be applied to collect calls. Does anybody knows if I can set this up in Asterisk? I'm using an E100P connected to the PSTN and a T100P connected to a Zhone 100. Version: Asterisk CVS-05/30/04-16:28:04 thank you Oz

Hey all! I'm having problems trying to set up an ATA 186 with my Asterisk box. When I get the phone to place the call, I type the extension and I only get busy signal after 5 seconds. So I can't call my Asterisk box from my ATA and either call from my Asterisk to my ATA. Does anybody know what can be happing? Log is attached.. tks regards Oz -------------- next part -------------- >

Hi, We currently have an Asterisk box (P4 2.4 Ghz, 512Mb RAM, 1 T100P, Zhone Channel Bank and 1 E100P) connected to an ISDN running without any problems. That machine is working for about 1 year. Two days ago, we decided to switch that machine for two PowerEdge 600SC (HA) and we got some problems. The running machine (P4 2.4Ghz) has an Intel motherboard with four 32-bit 3.3v PCI slots,

Hey All, I need to forward an extension to an other depending on the current time but I could not get it done with GotoIfTime. What I'm trying to do is ring on the extension 1 if time is between 8:00AM and 2:00PM and on extension 2 if is between 2:01PM 11:00PM. exten => 111,1,GotoIfTime(8:00-14:00|*|*|1-12?333) exten => 111,2,Dial(${Person1}) exten => 111,3,Dial(Hangup) exten

Hi all! Guys, I'm testing the Cisco ATA 188 with my Asterisk (version 09/10/03) and I faced with a low call volume heard just for people who is not "under" the ATA. I mean, if I call a person whose extension is connected at the ATA, he can hear me perfectly, but I get a low call volume. Is it possible to change this call volume? Can I do something in order to get a little high

I have a Cisco ATA-188 with two POTS phones and latest stable cvs. In any situations with call waiting (existing connection and calling again) the second call cause both calls to drop. This is the same for "internal" extensions and from external (ZAP and SIP). It seems to be a "502 - The transaction could not be executed, because the endpoint does not have sufficient

Hi All, Having searched the archives, I can see there has been much discussion at various points regarding capture of caller id information from good old BT. If I understand correctly, it seems that not only do the drivers not currently support it, but my X101P possibly/probably can't do it anyway due to hardware? So, that leaves me with the modem route, which seems more and more unlikely,

Hi I migrated to dovecot 2 and I started using lmtp for mail delivery. But I have found this on my logs "save failed to INBOX: Not enough disk space" . There is enougth space on disk. and I am not using quota plugin on lmtp my lmtp conf is as follow: # ## LMTP specific settings ## # Support proxying to other LMTP/SMTP servers by performing passdb lookups. #lmtp_proxy = no # When

Hi, I am compiling dovecot and I found dnotify & inotify. What does dnotify or inotify do ? do i need it. Thanks

Hello all, I am testing postlogin script with dovecot 2. I works but i would like that the imap and pop session get executed even if the post-login script fail. my postlogin script is very simple: #/etc/dovecot/conf.d/10-master.conf service imap { executable = imap imap-postlogin } # service imap-postlogin { executable = script-login /usr/local/bin/postlogin.sh unix_listener

Marcel Moolenaar wrote am 17.03.2008 17:28 (localtime): > > On Mar 17, 2008, at 3:33 AM, Poul-Henning Kamp wrote: > >> phk 2008-03-17 10:33:23 UTC >> >> FreeBSD src repository >> >> Modified files: >> sys/sys ata.h >> sbin/atacontrol atacontrol.8 atacontrol.c >> sys/dev/ata ata-all.c ata-all.h

Me again... I have service with a company that does not allow for a BYOD plan. They will not give out credential or server info either. Is it possible to run the FXS port of the ATA to an FXO port in *? The service I have is throug Broadvox Direct using the Mediatrix 2102. I have tried this using loop start and kewl start. The * box sees the incoming ring, picks up and starts my dial plan. But

hi all We have an ocfs2 cluster compose of 4 Debian lenny serveurs wich have access to an ISCSI LUN we have create a partition on this Lun and formated this partition as OCFS2. 2 serveurs does mail delivery (SMTP) and the two other are pop/imap servers. The smtp servers use dovecot LDA to deliver to mailboxes. La version de dovecot is 1.2.13, ldirectord balance conexion to the

Hi I wonder how can I know if a mailbox is not beeing consulted. I mean I have severals mailboxs but I know some account are not used. I like to know how to determine wich mail accounts are not beeing used. Is there any way in dovecot to know the las time a user loged in to dovecot? Thanks