similar to: sip to sip ?

Looking at my log file I found the following error: May 2 12:00:45 debian kernel: Zapata Telephony Interface Registered on major 196 May 2 12:00:45 debian kernel: No ISA tormenta card found at d0000 May 2 12:00:45 debian kernel: Zapata Telephony Interface Unloaded May 2 12:00:45 debian insmod: /lib/modules/2.4.20-8smp/misc/torisa.o: init_module: Input/output error May 2 12:00:45 debian

Hi All, I am a newbie to Asterisk. Presently I am working with Asterisk 1.4.17 and using it to make SIP calls. I have a configuration of Asterisk which serves the users in a particular domain, say internal.com I would like to make a SIP call from bob at internal.com to charles at external.com I have added the following lines in extensions.conf exten =>

Can anyone inform me if voip can be used on a IPv6 network? Does any hard phones/soft phones/Asterisk support it? Google told me that there was/is a bounty on it, but that expired august last year. Furthermore, there used to be a patch (Bernhard Schmidt), but that one is about a year old. I presume it can't be used on recent versions of "*" Hans -- pgp-id: 926EBB12

Hi, Asterisk Version : 1.2.15 Card : TDM11B (1 x FXO , 1 x FXS) I have internal dialling working okay SIP->ZAP (analogue phone) and ZAP (analogue phone) -> SIP. The problem comes when I try and make a outbound call. Here is my extensions.conf :- Code: [incoming] exten => s,1,GoToIfTime(17:00-09:00\mon-fri\*\*?outofhours|s,1) exten => s,2,GoToIfTime(*\sat-sun\*\*?outofhours|s,1)

Hi, We have tried upgrading from 1.6.1.14 to 1.6.2.9 and have found that we are unable to URI dial our clients. We run a multi-tenant server and have set sip.conf to forward calls to a public context based on incoming domain name. This was all working before but not it is complaining of a loop back as the source and target server are the same. Any ideas on how to overcome this problem as we dial

Hi The subject line says it all how do I enable this style of call. Pointers to the dns setup and asterisk setup would be great or even search words for google, as I am not sure how to search for this type of request. Alex -- There is no instance of a country having benefited from prolonged warfare -- Sun Tzu - The Art of War -------------- next part -------------- A

Hello, all. In our multi-tenant environment, we would like to be able to use the reinvite media redirection within Asterisk for calls within a tenant but not between tenants. We would like inter-tenant calls to be fully proxied by the Asterisk server. I think the answer is, "we can't," but I thought I'd ask anyway. I'd dearly like to remove the substantial traffic

Hi I noticed in a post that Timo might start work on a LDA Great! If so, here is something I would really like included. Support for PAM There is quite a lengthy reason behind this. In my organisation, user accounts are created on ldap. It is up to the services to create home directories whenever the user first uses them. For instance, Samba will do this the first time they connect to their

hi all i'm getting the below error when trying to compile asterisk-1.4 on redhat-9.0 any suggestions ? make[2]: Leaving directory `/usr/local/src/asterisk-1.4.0/menuselect' make[1]: Leaving directory `/usr/local/src/asterisk-1.4.0/menuselect' Generating input for menuselect ... menuselect/menuselect --check-deps menuselect.makeopts Generating embedded module rules ... [CC]

Is there a way to pass variables/arguments to the h extension ? for example : [default] exten => _1098933X.,1,NoOp(CARRIER TWT->TIM, EXTEN: ${EXTEN}}, SIPCALLID: ${SIPCALLID}, SIPDOMAIN: ${SIPDOMAIN}) exten => _1098933X.,2,SetVar(_PROVA="bla") [lot of stuff, agi, goto, tricks and magic that happens] exten => _1098933X.,10,Dial(${CHAN_DEST},,L(3600000:3599900)) <-

Hi, Is it possible to pick the context of a call from chan_sip based on the domain of the To: header of the INVUTE? I've had a quick look throught he code and can't see anything, I want to use the voicemail virtual hosting with chan_sip. Can the sip domain be picked out with a global in extensions.conf? This woud also solve my problem. If not is there any specifc reason/restriction

Hi all I'm after a Digium card that will allow me to connect an Asterisk box to.. 2 x sip providers 1 x company PBX 1 x POTS provider. Can anyone recomment a card that can do the job.

On 11/09/2020 18:30, bobby wrote: > I am now running 2.3.11.3 (502c39af9), and am still getting these > messages. OK, good. What is your current version and configuration (output from `dovecot -n`) Anything interesting in the logs? Any idea which deliveries are causing this? Can you obtain an LMTP protocol log for such deliveries? Regards, Stephan. > On Fri, Sep 11, 2020 at 10:52

hi all, how to establish a call between two asterisk servers for the sip users registered for the servers. ----- Original Message ----- From: <asterisk-users-request at lists.digium.com> To: <asterisk-users at lists.digium.com> Sent: Sunday, February 10, 2008 11:30 PM Subject: asterisk-users Digest, Vol 43, Issue 30 > Send asterisk-users mailing list submissions to >

dovecot-0.99.10.5-0.FC2 I have had a report from a user that when accessing their inbox, the connection is suddenly dropped. Dovecot reports the following:- imap(user): file buffer.c: line 357 (buffer_set_start_pos): assertion failed: (abs_pos <= I_MIN(buf->used, buf->limit)) From searching the list, I found an indication that it is a problem with an individual message. As the user

Hi all, I've been pulling my hair out for two days over this problem... I did *a lot* of Googling around, I searched the list archives to no avail - no one has the same problem! I have two Avaya 4610sw phones. I installed the latest SIP firmware using the TFTP server. So far everything looks good. Each time the phone boots, it retrieves the 46xxsettings.txt from the TFTP server. My problem

I use client Sjphone which work fine but i have Sniff a traffic.. - Sjphone send packet with OPTIONS to Asterisk - Asterisk ask with 404 not found This sequence come back often in my log. I don't understand why Sjphone Sens OPTION, and 404 not found.. Thanks for your help

Hi all, For those of you people that use Agents (with Agentlogin, not AgentCallbackLogin) on a call center, I have this need: when the agent logs in, a channel keeps running all the time that the agent is logged in to receive the incoming calls. How do I know which agent logged in (code)? Right now, if I query the login channel, there is no information about which agent is logged on: #

I have come across a strange situation, involving new accounts. I am running Dovecot 0.99.10-5 on Fedora Core 2. Dovecot is configured to use pam for authentication, and the dovecot pam service is configured to use pam_mkhomedir. When an account is created and the user logs in through SquirrelMail, there is no problem. The standard folders are available - Inbox, Drafts, Sent Items and Trash.

ok, stupid question time.... how does one use quotas and samba, i haven't seen anything mentioned in any man page re quote support.