similar to: Dialing from E1/T1

Hi all, I've tried search this problem on the list... no luck... The case is: without externip/localnet config on sip.conf [general] my SIP trunk works, but with no audio NAT problem (asterisk sends the private 192 address to the outside...) when I configure externip/localnet correctly my SIP trunk simply disappear! Checking the signalling with tcpdump shows me that Im sending the

Hi, Does anyone had the problem of asterisk SendFax + spandsp sending only the first page of a multi-page TIFF file? Seams to be related to spandsp ECM config. Any thoughts about it? Thanks, Gabriel -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20121005/ac471600/attachment.htm>

Hi all, Someone know how can I check for available members on a queue Before I queue the call, so I can do something else with it? Note that is not the case for joinempty Thanks, Gabriel Ortiz -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090718/462b725b/attachment.htm

Can I activate/deactive the canreinvite SIP flag on the dial plan? The idea is to allow reinvite only for exten <-> exten calls, and not for outbound calls -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090117/a53f3178/attachment.htm

Hi all, Does anyone uses astDB for a large amount of data, in special for implementing black lists with millions of numbers (i'd like about 2 or 3 million)? That would be held in memory right? Is this (memory consumption) the only problem I could face? Att. Gabriel -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I've noticed an unusual behavior on the dialplan execution: assume this DP: exten => _6XXX,1,NoOp(test1) exten => _XXXX,1,NoOp(test2) exten => _XXXX,2,NoOp(test3) If I call 6000 then test1 and test3 NoOps get executed, even though the pattern is different. I've always thought that if I call 6000 it would match the 6XXX pattern, that only has 1 priority, that would get

Hi all, After * starts the command "queue show" would not show any of the realtime queues, but just the ones that are in the queues.conf file. In this state de AMI would not send any "QueueMemberStatus" for that queues until a call is received by that realtime queue. Anyone knows any whay to load this information in *'s memory without the need of the queue receiving a

Hi all, When I have 2 masks that would like to execute the same logic, there is the way to use the Goto (or any other) command without changing the ${EXTEN}? Eg. DID range is 1200-1349 -> call Macro(disca), what mask to use? (I just got it with 2 masks, but I didn't wanted to duplicate the dialplan for both) [test] exten => _12XX,1,Set(DIR=3) exten =>

Hi all, I'm trying to use the CDR() function on the "Variable" header of the Originate AMI action, but it isn't working. Anyone knows anything about this problem? asterisk 1.4.26 Thanks, Gabriel Ortiz -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, A question on channel originating (call files and AMI Originate): How can I change the CALLERID(num) var (because of the E1 provider needs), but having another n?mber (the original one) stored on the "clid" CDR field on the database? A channel agnostic solution would be the best one, without having to deal with the problem based on what type of Tech used for the outgoing

Hi all, I'm trying to send an email with the voicemail details and I want to send a HTML link on it to make a click2call to the voicemail main, but the email is send with 'text/plain' encoding and thus it will not show the link, but the HTML in plain text on the body of the email, How can I change the enconding to 'text/html' so the link will get displayed correctly?

Hi, first i will explain my experiment : We have ESX server running virtual machines .We want to migrate to XEN. As i did not found any tool for transfering ESX to for example real server, or better to XEN --> please correct me if i am wrong , i found following approach : i will inside virtual machine on ESX boot from Live-CD of CentOS , using dd i will transfer images trought scp to CentOs

Hi, first i will explain my experiment : We have ESX server running virtual machines .We want to migrate to XEN. As i did not found any tool for transfering ESX to for example real server, or better to XEN --> please correct me if i am wrong , i found following approach : i will inside virtual machine on ESX boot from Live-CD of CentOS , using dd i will transfer images trought scp to CentOs

Hi all, How can I playback a file within an active call? I've tried with ChanSpy whisper mode like this (using AMI): Action: Originate Channel: Local/9999 at default Priority: 0 Variable: MSG=test Application: ChanSpy Data: SIP/1234-123 Async: 1 and in the dialplan: [default] exten => 9999,1,Answer() exten => 9999,n,Wait(2) exten => 9999,n,Playback(${MSG}) Where

Hi , i am facing a strange problem. I have centos , i wan to access svn trought apache using mod auth ldap. This is what i have configured AuthLDAPBindDN cn=svn,ou=Operators,o=Organization AuthLDAPBindPassword Pass1 AuthLDAPURL "ldap://ldap/ou=Users,o=Organization?uid" AuthLDAPGroupAttribute member AuthLDAPGroupAttributeIsDN on Require

Hello, First, I'd like to say sorry if this isn't the right mailing list to ask my question. I'm writting some application(on Linux) wich relys on sshd, so I thought there is nothing wrong in posting my mail here. The aplication i'm writting should, at some point, kill the user from his terminal line. To achieve this I read the users utmp entry and kill the ut_pid in it.. It

Hello, I cannot call my server over the internet with SIP anymore. Even when I do a maximum logging on my firewall, I don't see packets coming from outside. I've tried it from an ekiga.net account and an sip2sip.info account. What could be wrong? I would expect incoming traffic on port 5060 UDP... The account is "paul at vandervlis.nl". This should connect trought DNS to the

Hi to everyone! I have succerfully instaled my new Asterisk 1.4 on my debian etch. I have my users in sip.conf like this: [200] type=peer host=dynamic context=home secret=200 callerid= 200 dtmfmode=rfc2833 nat=yes mailbox=200 at home disallow=all allow=ulaw I can make calls in my LAN but i can?t ear comunications with another client trought Internet. My Asterisk is in my LAN and i not have a

Hi, I need to connect 2 asterisks in 2 different countries (A and B) for one company so it's possible to make connections between the 2 offices. For connectivity reasons (NAT traversal) i want to connect the 2 asterisk with IAX so that when a user on office A connects via SIP to user on office B the call is going trought IAX channel. Can anyone give me an ideia how to accomplish

Hi there, i''m trying to shape incoming trafic on my linux box. I want it to depend on the source IP adress of the packets (in fact, i got a VPN on an ADSL link and wanted to get at least/max 64Kbit/s for the VPN) So i tried this : tc qdisc add dev ppp0 root handle 1:0 cbq bandwidth 512Kbit avpkt 1000 cell 8 tc class add dev ppp0 parent 1:0 classid 1:1 cbq bandwidth 512Kbit rate