similar to: Can asterisk record the duration of users putting on hold?

Hello Team, Can you help me on this? I have attached here the queue logs of my asterisk. I've searching a parser for this. I do not know what are the meaning of that logs. Thank you so much. Your response is highly appreciated. Regards, Cristina -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, I have installed X-lite client on a windowsXP machine and asterisk on an enterprise linux m/c. The client is sending a registration message to asterisk server. It is able to process the message and sends 200 OK back. But later it says "Scheduling destruction of sip dialog xxxx ". Then it says "Really destroying sip dialog xxxx". What to do for this problem??? I

Hi all, I am using asterisk1.2.14,realtime and I find there is a strange case in the receiver's display. I have a dial plan to route a call to the destination. I haven't set the callerid(num) for the caller. In the receive ends, it's display shows "asterisk" when I make a call to the receiver. I wonder why "asterisk" shows in the display as I haven't set

Hi, I've been doing some testing on moving from 1.2 to 1.4 and one issue I've encountered is re-transmits whenever an INVITE is cancelled. I have a stateless SIP proxy in fron of my asterisk servers (all it does is direct requests to one asteisk server or another) and the re-transmits do not occur on 1.2.17 which is the current verion I have in use on my production servers. The

I have a new asterisk install (1.0.7) - and in case it's relevant I'm not using autoload option in modules.conf. Generally all is working well. However, when I make a call from my softphone and try to leave a message, the message is cutoff after a few seconds (whenever I pause for 1 second between words). Strangely, when I use an analog phone connected to my ATA, I can record as long as

I set up my * server using its publc IP address. Now that i switch over to using the domain name, X-Lite can't log in. =========With Domain Name (doesnt work)============ Transmitting (NAT) to 85.250.206.46:6007 <http://85.250.206.46:6007>: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 85.250.179.93 <http://85.250.179.93>

Hi; I am running asterisk 1.6 from Fonality (Trixbox PRO). I am trying to initiate a call FROM a softphone client to asterisk (either an internal 4 digit extension call) or an outside line via a SIP trunk. In both cases, asterisk rejects the call with a 420. In this case, it?s a call from x3992 to x4415 Does this require a change on the softphone for x-call-detail? <--- SIP read

I have a SIP account that I can successfully register with XTEN and a Sipura-2000. I have yet to be able to get it to authorize with *. My XTEN looks like: Username: 001234 Password: xxxx Authorization Username: 001234 Domain: domain.net Register with domain:

The SIP trunking service that I am trying to set up keeps saying that my registration from Asterisk is invalid. Asterisk registration: REGISTER sip:{registration_ip} SIP/2.0 Via: SIP/2.0/UDP {asterisk_ip}:5060;branch=z9hG4bK5c2eb10c;rport Max-Forwards: 70 From: <sip:{registration_user}@{registration_ip}>;tag=as5579cc0c To: <sip:{registration_user}@{registration_ip}> Call-ID:

Hi list, How do I make Asterisk 13 (using PJSIP channel) to handle PUBLISH sent from the phones? The trace looks like: ## PHONE -> ASTERISK ## PUBLISH sip:1001 at example.com SIP/2.0 Via: SIP/2.0/UDP 172.31.19.250:2048;branch=z9hG4bK-w2orn21sre9u;rport From: "1001" <sip:1001 at example.com>;tag=98slbhbn16 To: "1001" <sip:1001 at example.com> Call-ID:

REPOSTED with MORE Info and Modified Subject Line: -------------------------------------------------------- I am using one of the Minute Provider to dial out USA numbers. Now in one of my process, we need to Dial IVR and the enter DTMF digit and then it connects to the automated IVR. When I dial out the IVR directly using Xlite and VOIP Mins provider , it works perfectly. but when In try from

Did a quick-n-dirty install of 1.1.alpha2 on a Fedora Core 6 server... am getting the following message in the maillog, though there seems to be no problem accessing mail. Error seems to be consistent for all users. Running 1.0.3 on the system previously. Aug 8 22:49:26 stage2 dovecot: IMAP(testuser): mkdir(/var/spool/vmail/192.168.0.199/.imap/testuser) failed: Permission denied This is what

Dear Dovecot mailing list people - What I'm trying to do is use Postfix (with virtual users) and a POP3 server (we don't want IMAP - don't ask) to work against our new Active Directory servers that will be coming soon. Before we roll anything out we're testing it in the lab. Here's what we've got: x86 server running Red Hat Enterprise Linux 3 Postfix 2.0.16 Dovecot-1.0

IVR Number :17275691533 When I try it from xlite configuring my provider directly, it works perfectly. When I try to dial out from dialer , it doesnt work. [sip8] type=peer username=user fromuser=user authuser=user secret=password host=8.14.146.111 nat=no canreinvite=yes insecure=very disallow=all allow=g729 allow=ulaw context=default dtmfmode=rfc2833 What cld be the reason ? --------------

We have an odd problem, where inbound H323 (chan_h323) calls will sometimes cause a Ringing Congestion that appears to keep the channels open and never release it until we kill and restart asterisk. These "Ringing Congestions" start to pile up, which eventually crashes Asterisk. H323 Gateway -> Asterisk (chan_h323) -> Tor2/PRI -> PSTN Has anyone ran into this problem or

I've read the docs. I've bought the book. I've searched the lists. For three days and still no joy. Please help. Here are the details: OS: Redhat 9.0 on a Dell Poweredge 2650 Using stable samba 3.0 rpm downloaded straight from the site Configuring by any means necessary: webmin, swat, but right now, smb.conf I have set up a linux account for smbuser, added him to ntadmins group,

Hi, I have been working on setting up a 4 replica gluster with over a million files (~250GB total), and I've seen some really weird stuff happen, even after trying to optimize for small files. I've set up a 4-brick replicate volume (gluster 3.13.2). It took almost 2 days to rsync the data from the local drive to the gluster volume, and now I'm running a 2nd rsync that just looks for

Hi, I was told to do so by configure of openssh-7.1p2 -:) ... configure: WARNING: sys/audit.h: present but cannot be compiled configure: WARNING: sys/audit.h: check for missing prerequisite headers? configure: WARNING: sys/audit.h: see the Autoconf documentation configure: WARNING: sys/audit.h: section "Present But Cannot Be Compiled" configure: WARNING: sys/audit.h:

Hello, really new to advanced linux routing, but I have managed to setup something that sort of works for my needs. I have a linux box with this configuration: Internet - eth0 - eth1 - many users I''m using a NAT to allow my users Internet access. On some users, I wish to shape their download speeds as to not overload our outside connection. Here is what I do to shape them at 64k

(2005.10.10, 14:25) I am running Samba 3.0.14a-2 on Fedora 4. I have no problem seeing the Linux file system from any of the Windows (2000, SP4) computers attached to the network. The problem is that smbd processes are created for no apparent reason and never die. I noticed this when the computer had been up for about three days and there were some 600 active smbd processes! What could be the