similar to: Rewrite Outgoing Number

Kevin Larsen - Systems Analyst - Pioneer Balloon - Ph: 316-688-8208 asterisk-users-bounces at lists.digium.com wrote on 12/14/2017 09:36:06 AM: > From: "basti" <mailinglist at unix-solution.de> > To: asterisk-users at lists.digium.com > Date: 12/14/2017 09:36 AM > Subject: Re: [asterisk-users] Rewrite Outgoing Number > Sent by: asterisk-users-bounces at

Hi :) I hope someone has a hint concerning Early Media. The situation: My Asterisk is connected to small local carrier who works with several SIP servers. I traced some SIP headers and find something like this: Via: SIP/2.0 UDP sip1.provider1.de In the SDP part I found something like this: o=- 2268929 0 IN IP4 sip2.provider1.de c=IN IP4 sip2.provider1.de If I send

I am using GotoIf all over the place in 1.4.8 but for some reason, the following in my dial plan: ############################################################# exten => _1NXXNXXXXXX,1,GotoIf([${EXTEN} = "15554441212"]?100) exten => _1NXXNXXXXXX,n,Dial(SIP/provider1/${EXTEN},60) exten => _1NXXNXXXXXX,n,Dial(SIP/provider2/${EXTEN},60) exten => _1NXXNXXXXXX,n,Hangup exten =>

Greetings list, Over the last few days I've been gearing up to replace a couple of our servers with 1.6 as something of a testbed, but I'm encountering a few problems, and wondering if anyone can help... In extensions.conf, there are a number of contexts defined for each group of users, along the lines of: [groupa] [groupb] etc. In each of those, there's a command include =>

I need to create a legend for a simple scatter plot in the following format. This is Blah1 number1 number2 This is Blah2 number3 number4 . . . This is Blah6 number11 number12 I looked up these help pages and found the following solution. lStr<-c(Blah1, Blah2,....Blah6, number 1, number2, ...number12) legend(x="topright",lStr,ncol=3) So this creates the tabular format I am

Greetings all, I'd like to start implementing a private DUNDi peering group between one of our asterisk servers hosted at a datacentre and the various asterisk boxes sitting at clients' premises. On most of the clients' boxes the dialplan will have an [in-pstn] section containing the various numbers that should be recognised by that box. Where they're from a VoIP provider they

Hello everyone. I've stumbled upon a strange networking issue with multiple interfaces on CentOS 5. The network setup is just like the diagram in http://lartc.org/howto/lartc.rpdb.multiple-links.html It looks like linux is not routing correctly outgoing packets on interfaces different from the one of the default gateway, but instead broadcasts an ARP request on the link, looking for the

Hi, I have two X100P cards in the same system. I can use both of them to initiate and/or receive PSTN calls. I want now to define separate context for each of them, in oder to route inbound calls to different extensions. This is what I have now in zapata.conf file: [channels] language=en context=inbound-analog signalling=fxs_ks usecallerid=yes callwaiting=yes echocancel=yes

Hi, I am having trouble passing variables via the call files, here is my call file via the php: fputs($oSocket, "Action: login\r\n"); fputs($oSocket, "Events: off\r\n"); fputs($oSocket, "Username: $strUser\r\n"); fputs($oSocket, "Secret: $strSecret\r\n\r\n"); fputs($oSocket, "Action: originate\r\n"); fputs($oSocket,

Hello, I am calculating 'Betweenness' of a large network using R. Currently, I have the node-node information (City1-City2) in an excel file, present in two columns where column A has City1 and column B has City2 that city1 is connected to. These are the steps that I go through to calculate betweenness of my network. a) Convert the City1-City2 (text) into Number1-Number2 in the excel

Hi guys, I have an asterisk box with sip 20 internal extensions and 100 lines registered on a external voip provider. For most part of time, it work fine, but in few moments it act ignoring sip packets becouse my ip phones can't register in asterisk and asterisk can't register his 100 lines in external voip provider. I have log's only for external registration error: [Apr 16

Hey everyone, I have a more of an opinion question then a technical question. The asterisk server I am setting up is going to host 3 different businesses. Each business is in the same building, and on the same network. My question is regarding calls coming in and going out. We are a small ISP and have a lot of numbers that are forwarded to our phone system. The other companies have about 3

Hello In the LARCT how-to subitem: 4.2.2. Load balancing the following phrase says: "" Instead of choosing one of the two providers as your default route, you now set up the default route to be a multipath route. In the default kernel this will balance routes over the two providers. It is done as follows (once more building on the example in the section on split-access): ip

Hi, I put on sip.conf the following line #include "sip.d/*.conf" inside I have files like that provider1.conf provider2.conf But asterisk does not want to load it This is the error Sep 1 13:18:35 VERBOSE[8756]: == Parsing '/etc/asterisk/sip.d/*.conf': Sep 1 13:18:35 VERBOSE[8756]: == Parsing '/etc/asterisk/sip.d/*.conf': Not found (No such file or directory) this

Ok ive been trying to get this to work for about half a year now. Ive searched all over the internet for a solution for my problem. Ive found some solutions, but they only led me to yet more problems. What we want to do is the following: I live in a student complex with 7 other people. Every room has its own internet connection from the same ISP. Ip, gateway, subnet are asigned through dhcp on

Hi! I'm really hope you can help me solve a little mystery, the mystery is probably just my misunderstanding ! sorry... I've got an iaxy talking to my * box which connects to two providers. I'm running the stable release of the pbx. The only thing is that when dialling from the iaxy the ringing tone isn't heard while calling someone - you just hear silence then, they either

Hi, I''ve been experimenting with ip route for the last few days to get load sharing accross 2 providers working. While it works most of the time, on a few occasions, packets are routed to the wrong interface. I''m not sure to understand rules and routes traversal correctly (I couldn''t find answers in the howto). So, here are my questions: 1. How does the rule

Hi, I have two loosely related questions which could make my live again a bit easier: 1) Is there a simple way to select a range of columns in a data frame using column names? I am thinking of something like mydf[1,"col4":"col8"] 2) I have a data frame with many columns and they all have short variable names which is good in most cases but sometimes it would be nice to have

Hi I want to be able to use one provider if I dial 0 before the number and another if I dial 1 before, how can I do that in asterisk 1.6? /ralf Ralf Tr?skman, IT AdLibris AB, Box 3667, 103 59 Stockholm. Bes?ksadress: Sveav?gen 56C, 111 34, Stockholm - Obs ny address! Dir: +46-(0)8-5460 60 91, mob: +46-(0)70-7548074, vxl: +46-(0)8-5460 60 00, fax: +46-(0)8-5460 60 99 ralf at

Is there a way to override the fromdomain specified in the sip.conf and instead set the value from the dialplan? If we use: Set(CALLERID(num)=user at domain.com The SIP From header turns into: user at domain.com@10.10.10.10 We want user at domain.com, and we can't have an entry in sip.conf for every provider. -- Eric Chamberlain