similar to: Several mailboxes on SIP peer

Hello, I have a few entries for sip peers in sip.conf with different name and username, like [TestSIPUser] type=peer host=dynamic username=testuser secret=1234 context=test_context [TestNewUser] type=peer host=dynamic username=newsipuser secret=3456 context=test_context When a call is made from any of these peers I want to get the username of the peer. for eg:- If a call is being made from

Hi there. My problem, I can't figure out how to ask this question. So, hopefully someone out here can point me to the FM on this. I would like to have either a web page or an application that I can view that whenever a call arrives on the Asterisk server the application will display the callerid information. I've found quite a few examples of the reverse of this. To where a script is

I have a instance of Asterisk 1.2.14 that is being run from safe_asterisk. Asterisk is seg faulting and NOT generating a core dump. Why would that be? How can I make it dump core? Is there a setting in the safe_asterisk script that I am missing? Thanks, Doug. -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have setup Asterisk on 2 Fedora Core 8 machines, and have made it to record all incoming calls. One of the box that have Asterisk 1.4.18 is properly merging calls and the other box that has Asterisk 1.4.15 is recording the calls but not merging them, I have made sure that SOX is installed on the box. Here is the Dialplan of both the machines : exten => 1234,1,Answer() exten =>

Hi *, does someone has a directory that works with the Cisco 7960 and astdb or mysql/ldap? Regards, Andreas _________________________________________________________________ Gaming galore at http://xtramsn.co.nz/gaming !

I have a carrier who tells me he will be sending me traffic from a wide range of IP addresses. so I set up a realtime peer as follows: [peer] defaultip=xxx.xxx.xxx.xxx host=xxx.xxx.xxx.xxx deny=0.0.0.0/0.0.0.0 allow=xxx.xxx.xxx.0/255.255.255.0 insecure=port,invite Yes, he's really claiming to originate from any of the IP in the block When I leave the host blank, we reject calls with a

Hi All; How can I test manager.conf? Can I telnet to the asterisk machine at the port 5038 and send and receive commands to test if the manager is working fine? How? Regards Bilal

I've read a lot of conflicting information on this around the web, and wanted to see if I could get some thoughts for any of you.. What's the proper (or best, etc) build order for install Asterisk and it's needed libraries. Most often I see 1. Zaptel / Dahdi 2. libpri 3. Asterisk. However, I've see several references to installing libpri before zaptel is installed. Should

https://fedoraproject.org/wiki/Ovirt_Node_Spin I noticed that virt-manager-tui is in rawhide finally, but the package name implies that it's not going to be in f16? virt-manager-tui.noarch 0.9.0-4.fc17 rawhide Cole, is this intentional or just smth that we need to follow up on? Also, I've noticed that ovirt-node needs refreshing... iirc apevec did tag/release of 2.0.1 from

I've got an Asterisk server, and several SIP phones behind our router here. Things are working just perfectly inside the network, just as the should. However, I'm not trying to configure my asterisk server to talk with SIP services outside our network, once such example is my gizmo project account. This isn't working out to well. Would it be useful to have a SIP proxy outside of my

Greetings all. I''m using the Microsoft "NorthWind Traders" database "orders" table as test data for the Rico LiveGrid JavaScript. The problem I''m noticing is that some of the records in the "ShipName" field contain ampersands -- which for some reason are not being escaped by Rails. Rails does in fact automatically escape all of the other

Hi I have a new Samba 3.6.10 server running on Solaris 10. The server is a member of the local Active Directory (which I'll call "DOMAIN" in this email). Unix username resolution is via NIS. All domain users have NIS usernames as well.Winbind is running to allow SMBD to perform sid>uid mapping and I have setup idmap_nss. I am not using winbind in /etc/nsswitch.conf as NIS

Hi List, While I know that "upping" ulimit will fix the issue I am trying to understand what will cause it. I have a few set ups that are almost exactly the same yet some machines used to give this error often and others don't. I also noticed the error a lot more on my boxes running 1.4.X. TIA. Dovid -------------- next part -------------- An HTML attachment was scrubbed... URL:

Darryl Moore <darryl at moores.ca> schrieb: > I'd start by turning on sip debugging in asterisk > >sip set debug ip [your_phone_ip] Really destroying SIP dialog '490d1996593c8e11217828b71aae5c4d at 172.16.34.133' Method: OPTIONS Reliably Transmitting (no NAT) to 192.168.200.11:5060: OPTIONS sip:00493512222222 at 192.168.200.11:5060 SIP/2.0 Via: SIP/2.0/UDP

Hi, Is there any Royalty one needs to pay when using the inbuilt exisimg asterisk on hold music or when using any other mp3 from a music album. I think we need to pay for the later, but I am not sure if we need to pay for the inbuilt asterisk(freepbx) on hold music. -- Deepak --------------------------------- Yahoo! Answers - Get better answers from someone who

Darryl Moore <darryl at moores.ca> schrieb: > Ahh. Seen that before! That suggests to me that you don't have your > sip.conf records setup right. > > What's your sip.conf look like? Well, here what I wrote in my sip.conf: register => 00493511111111:MYSECRET at pbxluca/00493511111111 register => 00493512222222:MYSECRET at pbxfax/00493512222222 register =>

Dovecot logs as follows: Feb ?1 19:36:34 speedy dovecot: imap-login: Login: user=<rsmith>, method=PLAIN, rip=10.163.195.83, lip=10.163.195.82, mpid=7481, session=<EbuSsb4qPQCto8NT>Feb ?1 19:36:37 speedy dovecot: imap-login: Login: user=<rsmith>, method=PLAIN, rip=10.163.195.83, lip=10.163.195.82, mpid=7483, session=<qfSzsb4qQQCto8NT>Feb ?1 19:36:37 speedy dovecot:

Hello list, When you have an asterisk box connected between the VoIP phones and an PSTN gateway what is the role of asterisk. Proxy server: stateful or stateless? From what i read in the: "Understanding the SIP, second edition" from Alan B. Johnston i think that asterisk is a stateful proxy server as well as registration server. Am I right? Can asterisk be configured to work as

I'm using dovecot RC6. I have a group named "cmpymail" with 2 users "jdoe at arinbe.com" and "jsmith at arinbe.com". I set up a mail folders like: drwxrwx--- 4 cmpymail cmpymail 4096 2006-08-13 02:21 cmpymail drwxrwx--- 3 jdoe users 4096 2005-11-21 13:34 jdoe drwxrwx--- 3 jsmith users 4096 2006-08-13 02:27 jsmith drwxrwx--- 3 fred users

Good afternoon folks. Earlier today, I started upgrading a few of our servers to 5.4 based on input from the list. So far, all has gone well. I have about 6 servers (not very many, but still) that need to be upgraded. Instead of taking precious bits from the mirrors for each upgrade, I was curious if I could rsync the 5.4 directory from a local mirror, and configure yum to use that repo, if