similar to: externip nat audio sip trunk issue problem

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

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:

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, We have some Asterisk servers that we are moving behind a NAT to preserve public addresses and make room for growth. This is Asterisk 1.4 NAT works very good with the externip/localnet-setting when we are connected directly to our teleco. But when I try to use NAT and put them behind our Kamailio something interesting happens: The media-address in the SDP is the internal ip and not the

Hi all! Do anybody has a full working environment using externip on an asterisk box behind a nat? I tried with two diferent boxes (Elastix-1.4.24 e Trixbox-1.4.22-3)and the asterisk do not replace neither contact, neither sdp headers info with the externip informed on sip.conf general parameters. I used these two statements: externip=XXX.XXX.XXX.XXX localnet=192.168.200.0/255.255.255.0 Do

Hi All, Is it possible to specify more than 1 localnet? I know this is an odd question. I have a customer that has multiple sites linked by VPN. Main range is 192.168.33.0/24 and a remote site is 10.1.1.0/24 We want to allow some access to the public IP address at the main site. For this to work I need to use the externip and localnet directive. If I do this it rewrites the SDP with the

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

ekiga registration fails. I've set nat = yes ( also blank ) and i've set externip. Anybody have a sip.conf that works? Here's the sip debug: Reliably Transmitting (NAT) to 86.64.162.35:5060: REGISTER sip:ekiga.net SIP/2.0 Via: SIP/2.0/UDP 10.10.11.180:5060;branch=z9hG4bK17818198;rport Max-Forwards: 70 From: <sip:test at ekiga.net>;tag=as64618445 To: <sip:test at

Hi, A have an asterisk connected to a legacy PBX trought an E1 and to the PSTN trought another E1. When the legacy user dial to the PSTN the call pass trought Asterisk. All works OK, the only problem is the delay on the Asterisk server when it receives the digits from the 1st E1 link. It will only make the call when the digit timeout expires. Is there a way to make something like

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:

I am having difficulty configuring SIP behind NAT (using latest CVS). Using sip.conf: [general] port=5060 ; Port to bind to externip=ww.xx.yy.zz bindaddr=0.0.0.0 nat=yes register=>[userid]:[password]@voiptalk.org/2000 [voiptalk.org] nat=yes externip=ww.xx.yy.zz type=friend secret=[password] nat=yes reinvite=no canreinvite=no I fail to register. SIP Debug gives: SIP

Hi, I have the following scenario. I have an Asterisk server running on an internal IP address behind a firewall, and I have a remote user trying to connect to my Asterisk box behind his firewall, but he can't seem to get a connection. I have opened up the port (5060) so that he can connect through my firewall, but it still doesn't appear to want to connect. I am pretty sure that the

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?

Will this do the intended thing? This is in sip_nat.conf which is included in sip.conf: externip=192.168.0.200 localnet=192.168.0.200/255.255.255.0 externip=64.168.237.110 localnet=192.168.1.2/255.255.255.0 I have Asterisk running on a box with two Ethernet interfaces and bound to both. One interface, 192.168.1.2 services clients outside the firewall who are led to believe that Asterisk is

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

Dears; It seems my service provider is requesting a complicated settings to allow me to send from behind NAT. What they said: "It shouldn't matter as long as you are handling the NAT correctly your end. We do not fix NAT so if you're sending internal addresses in your INVITEs or SDP then things will fail but if you're handling it correctly, we shouldn't tell the