similar to: Not receving anything from the list

Since Monday I didn't see much traffic.

A regular Dial(somepeer) to a SIP peer which doesn't reply at all seems to not time out, or at least have a very long time out. We have a set up where we can dial two different peers, a primary and a backup peer. If the first one dies completely, so that no error messages (no ICMP unreachables or anything) are returned, Asterisk does not continue in the dial plan but just gets stuck on that

Dear All I have jus migrated to centos 5 and jus wondering why n where this problem could be server A which is running for the past 2 years used as my primary dns and mail server redhat linux 8 sendmail ver-8.12.5-7 bind ver -9.2.1-9 since the server is old and is prone to all security problems i have a new machine which i have installed the following and will replace my existing

Another "me too" (also Gmail). I just received my 4th "account suspended, too many bounces" email, after having several days of lost mailing list content over a short vacation break the last time. When I notified the admin email account of the failure, it seemed the responder missed the point about the emails, saying the link had expired (it had been more than three days

I think that limit was increased in 1.6.6 or 1.6.7. -----Original Message----- From: C F [mailto:shmaltz@gmail.com] Sent: Tue 10/10/2006 6:57 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: Re: [asterisk-users] RE: Welcome to the "asterisk-users" mailing list On 10/10/06, Eric ManxPower Wieling <eric@fnords.org> wrote: > What I

Hi there. I am using dovecot in almost all the mail servers I configure, but I have not noticed some problems till now, when I configured a server for a japanese language based customer. Now I know, the problem is not about this customer, but it seems to be a constant in all my servers (for some reason we did not noticed that before). The situation is like this. When receving mail with extended

We must have had the magic version of 1.6.x then, because we increased our buddy watch limit from 8 to 48 in that version. -----Original Message----- From: Jessee J Holmes [mailto:jholmes@atacomm.com] Sent: Wednesday, October 11, 2006 8:18 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] RE: Welcome to the "asterisk-users" mailing list Limit

Dear All I have jus migrated to centos 5 and jus wondering why n where this problem could be server A which is running for the past 2 years used as my primary dns and mail server redhat linux 8 sendmail ver-8.12.5-7 bind ver -9.2.1-9 since the server is old and is prone to all security problems i have a new machine which i have installed the following and will replace my existing redhat dns +

Hello, I have an issue with Asterisk 12 PJSIP. When receving an INVITE with FROM URI that has a port number, the Asterisk removes the port from URI on consecutive Responses / Requests. This causes an issue with one of our SIP servers (it doesn't recognize the response / request). Below you can see an incoming INVITE and the outgoing 200OK response. I have highlighted the issue in Yellow. Does

Can anyone recommend a call accounting solution with rating for post paid billing that works well with asterisk using the account code or any other info from the CDR? I don't want the billing software to any phone calls for me, therefore any solution that modifies my extensions.conf is out, nor does it have to allow for customers the ability to log in to check their usage/balances. I have

Hi, dear dovecot-users, is it possible to make dovecot ignore the DELETE command for some accounts? Basically what I want to achieve is: users shall not be able to delete their emails - but they still should be able write emails. This is meant for a setup where different users are sending and receiving emails for a "support" account. I looked into IMAP virtual folders and ACL but I

Hi,all I have a POP3 problem about multi users from one IP address concurrent. [Issue description] The network architecture is as following. I have setup a mail server(sendmail/dovecot) in PlaceA with IP_A. And about 6 users are receving mail from the same IP_B( NATed public IP) in PlaceB. Their mails are many( about 1Mb/hour each) and ususally set fetch time as 10min/fetch. Then they

Hi, libvirt users and gurus, I am a newbie to visualization, hope this question is not too silly, please bear with me. I had installed three guest on Fedora 21 with the virt-install tool, by: virt-install --name=f21_x86_64_guest --file=/var/lib/libvirt/images/guest.qcow2 --cdrom /workStorage/work/down/f21/Fedora-Live-Workstation-x86_64-21-5.iso --ram=2048 virt-install --name=f21_x86_64_guest1

you think ? > > ** Login/Logout of queues, Day/Night mode buttons with indication (1.6 > has this as well). > ** Company internal directory on the phone updated on the PBX Some (most ?) IP phones support this > > ** System Speed Dial on the display updated by the PBX This one is interesting. I can't see a way to do it. Ant idea ? > > ** Call Fwd by PBX with LED

Hello my friends... Currently we are using the following firmware versions on ours aastra 55i: Firmware Information Attribute Value Firmware Version 2.1.0.2145 Firmware Release Code SIP Boot Version 2.0.1.1055 Date/Time Jun 20 2007 06:20:29 Can we make a firmware upgrade to the latest one: 6755i (55i) SIP, V2.5.3.18, January 2010 , English , ZIP , 2,849 KB on the site:

Is there anything like "nss_updatedb" [1] for ldapsam account information backend? nss_updatedb caches unix account information, so it is available even when the LDAP directory isn't available But ldapsam stores additional account information. How can I cache this additional account information, so it is also available even when the LDAP directory isn't available? [1]

Is there anything already out there that can efficiently convert a JSON string into Asterisk dialplan variables? Our current backend speaks JSON and we need to parse the response to construct the dialstring. -- Eric Chamberlain

Hi to all! I have this config, PSTN <--> AS5300 <--> ASTERISK I am using the Cisco as5300 to receive incoming calls and routing them to Asterisk for IVR. When I ran asterisk this is what I get when calling the voicemail demo. *CLI> -- Executing Playback("SIP/-081058b8", "transfer|skip") in new stack -- Executing Macro("SIP/-081058b8",

Hello! I'm tryuing to bring up an R2 connection but eventhough I've followed the guidelines in: http://zarzamora.com.mx/asterisk/17 something seems to be missing. When an incomming call is generated I get: Nov 24 06:01:17 WARNING[-197416016]: chan_unicall.c:612 unicall_report: MFC/R2 UniCall/24 <- 0001 [1/ 1/Idle /Idle ] Nov 24 06:01:17 WARNING[-197416016]:

Am I the only one having trouble with this list? Since the begining of the week I have not been receiving mail from the list like I used to, is this a gmail problem? or is it subscription problem? or is something wrong with the list? anybody else using gmail having any problems?