search for: estani

Hello, In the past I was in a situation that given an email UID I wanted to know which file in the Maildir directory this UID was saved. I did a small parser for $HOME/Maildir/dovecot-uidlist where, if the file had: 1000 W1838 S1796 :1557351041.10447_1.example I would ask for the UID 1000 and would return :1557351041.10447_1.example (and I knew that it was in $HOME/Maildir/new) Is there any

Hello, I'm checking some Billing Software for Asterisk. In opensource I only know (the name, I haven't used) AstBill. What other software should I check with similar capabilities? Thank you! -- Carles Pina i Estany GPG id: 0x8CBDAE64 http://pinux.info Manresa - Barcelona

Hello, We would like to have a conference with 15 users aprox. We think that Pentium 4 3GHz and 1GB of RAM should be enough. Only Asterisk running. We wonder if somebody has some other experience, good or bad. We will use Asterisk 1.2 (it is a small and short project for only this). Thanks! -- Carles Pina i Estany GPG id: 0x8CBDAE64 http://pinux.info Manresa - Barcelona

Hello, (I have asked it some time ago in Asterisk-es mailing list, so excuse me if anybody receive it twice.) I am trying to send SMS in Spain using landlines. It seems that app_sms.c only handles Protocol 1, but Spain and Italy are using Protocol 2. I have been searching in Internet without any results... anybody is sending SMS from Asterisk (or any method) using Protocol 2? (so, it seems,

Hello, Apologies for the extra long email. It's a situation with a solved problem and I might need to solve it again, I'd like to have some feedback. Two years ago, in a trip around the Antarctic with a very limited communication I deployed a mail system based on Postfix+Dovecot. I need to do something similar again and I used a "hack" that I thought that I might avoid (that

Hello, I would like to have "whisper" channel spy (not private whisper) in Asterisk 1.2. I see here: http://www.the-asterisk-book.com/unstable/applikationen-chanspy.html That is only available for Asterisk 1.4. I wonder if there is any way to emulate it in Asterisk 1.2. For example, to "join" two calls: one to use a private whisper and other one to use a normal chanspy.

Hello, In a couple of Asterisks, after type "sip show channels" we have a lot of these: IP_PEER dst_number something 00102/00103 unkn No (d) Rx: BYE IP_PEER dst_number2 something2 00102/00103 unkn No (d) Rx: BYE We are using ASterisk 1.2.x When I say "a lot" I mean more than 180, more than 230, etc. Is it normal? How we can remove it? Thank you very much, --

Hello, I'm sniffing traffic between Asterisk and a Softswitch. I see that, in "Decline" SIP packages, there is a header called "Reason" and I would like to access to the content of this header from Asterisk. How I can access to Reason header content? I would like to access here using ASterisk 1.4 and 1.2, but if it's only with Asterisk 1.4 will not be a big problem.

Hello, Fast question: from DomU I cannot ping Dom0, I only get 19:17:33.573370 arp who-has 192.168.10.1 tell 192.168.10.150 19:17:34.573421 arp who-has 192.168.10.1 tell 192.168.10.150 using tcpdump in Dom0. Why? --------------- Detailed question: I''m setting up a Xen virtual server. The environment is: -HVM -AMD64 -Debian Etch and using Xen from Debian repository (so Xen 3.0.3,

Hello, I wonder how Asterisk scales when we increment the Core's or CPU's of one computer. I see that Asterisk is only one process (I guess that it uses threads). But because Asterisk is only one process, this process is always executed in the same CPU. So we can have a 8 Cores server, with one Core running Asterisk, another Core running operating system stuff/other small daemons and 6

Hello, Today we setted up a server that needs to use MeetMe but doesn't have any Zap hardware. So we need to use ztdummy (at least, this was our idea). Rarely: zttest is not working at all (100% bad, using zttest -v doesn't give anything, etc.). Of course, after load ztdummy, there isn't any background or anything. It is the same kernel (Debian Etch default kernel, 2.6.18) than

On 9.5.2019 0.39, Carles Pina i Estany via dovecot wrote: > Hello, > > In the past I was in a situation that given an email UID I wanted to > know which file in the Maildir directory this UID was saved. > > I did a small parser for $HOME/Maildir/dovecot-uidlist where, if the > file had: > > 1000 W1838 S1796 :1557351041.10447_1.example > > I would ask for the UID

Hello, We have a machine (Debian Etch, Xen from Debian repositories, etc.) with two NICs: one for LAN (192.168.0.0/24) and another one with public IP address. After enable Bridging (in /etc/xen/xend-config.sxp): (network-script ''network-bridge netdev=eth0'') (vif-script vif-bridge) # eth0 is the LAN NIC I have eth0, peth0, vif0.1 (I guess that it''s usual for you) and

Hello, I am new user of Asterisk. Yesterday I was trying to call from softphone to Asterisk, and that Asterisk routes this call to sipphone.com provider. I have found information on internet about how to register to sipphone and it seems that I have done. "sip show status" (or similar command) in CLI was showing me that I was registered. To call was not working, and on Asterisk's

Hello, We are using this Asterisk: 1.2.14-BRIstuffed-0.3.0-PRE-1y Everything works fine but we have an issue (not often, but one call every some hundreds) I sniffed the communication between phone, Asterisk and softswitch. I can see that Asterisk receives a Cancel from phone but Asterisk never sends a Cancel to Softswitch. This makes us some problems: billing system doesn't allow next call

The client is travelling much of the time. Is there some way that he can use Port 80 so that the firewalls that he is behind won't block the connection? Any other hints or suggestions are very welcome!

Hello, We have an Asterisk server receiving calls using G711 (ulaw). This server has rerouters de calls to other server using G729 (we bought the codecs, installed, sip show channels shows the codec properly, etc.) Using G729, there is a "click" while talking. Well, more than a click it seems that voice is missing during some ms (maybe 100 ms?) Using G711 we don't have any click.

Hello, I'm receiving some traffic from a Softwitch to Asterisk When I'm hiding the CallerID in the softwitch, everything is all right. When I allow to send the callerid from softwitch to Asterisk (actually, I would like to have it) Asterisk rejects the call with a 407 Proxy Authentication SIP packet. I copy-paste the SIP Invitation: ------------------ Session Initiation Protocol

Hello, I am a Samba user, and I have been using long time without any big trouble. Last week I was working with a friend, using Fedora and Samba 3. Excuse me, I don't know exactly which Samba version. After some problems, we shared a directory (for example, /data). This works, fine. Then we created two new subdirectories: /data/a /data/b Then we access to shared resource using smbclient,