similar to: Call from internal num. to VoIP gate

hey * folk, am trying to configure a WellGate 3504A FXS SIP ATA (http://www.welltech.com.tw/products_ea01.htm) with asterisk. i've set up two SIP clients in sip.conf as follows: [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind SIP channel to context = default ; Default context for incoming calls [1235] host = dynamic secret = somepass context = default type

Skipped content of type multipart/alternative-------------- next part -------------- Nov 21 14:17:47 VERBOSE[32580] logger.c: <-- SIP read from 172.25.103.222:5060: INVITE sip:1715;phone-context=exp_net.ascom@ascom.be:5060;maddr=172.25.96.48;transport=udp;user=phone;x-nt-redirect=redirect-server SIP/2.0 From:

I am getting the following information and asterisk 1.0.7 is not answering the call. Any ideas? jerry ------------------ Sip read: INVITE sip:2828;phone-context=cdp.udp@qg.com:5060;maddr=161.49.198.102;transport=udp;user=phone;x-nt-redirect=redirect-server SIP/2.0 From: <sip:3173241052;phone-context=+1@qg.com;user=phone>;tag=c22da8c0-13c4-429efa71-657b8da-2ed To:

All, I am connecting to a CS 1000 nortel PBX. I can call out, I have limited success with call in. I get debug traffic that a call is coming in but I get the message "Unable to create/find channel". I was expecting that incoming calls over the trunk would be handled from my sip definition and goto the nortel context. It is not. Below is the actual incoming call debug information. I am

Hi eveybody again! I don't want to be annoying, but if nobody can help me with this, I'll have to desist of working with SIP.I have some questions about SIP, as I wrote in another mail. I have a SIP Gateway and I have two phones (an analog one and a DECT one) conected to it.Also, I have two Dlink dg102s with four phones conected to them. The main problems are two. Calls between the

Hello, I am working with 1.8.32.2 which I have patched with t38-gateway and PRACK. t38 is tested and working fine with Zoiper client but I can't get the t.38 software from Biscom (FAXCOM) to talk. In my first attempts I found FAXCOM announces that it supports 100rel so I added the PRACK patch hoping that would do the trick. Now it gets a little further but * complains about rejecting a

I am attempting to get a planet VIP-150T to register with asterisk 1.2.4. After searching google I've found what appear to be instructions in German, Russian and Spanish. Has anyone perhaps seen this before? Asterisk is kicking back the following error: Feb 14 09:59:32 NOTICE[21765]: chan_sip.c:10851 handle_request_register: Registration from '<sip:101@192.168.100.240>'

Here is the CS1000 log. Again, the CS1000 using SIP accepts incoming calls just fine. However, using outgoing call files the CS1000 is hanging up after I answer the call. I dont know why? Thanks, for any assistance. Jerry my sip.conf entry is: [Nortel] type=friend dtmfmode=rfc2833 username=XXXXXXXXX disallow=all allow=ulaw allow=alaw

I'm trying to debug a NAT issue and I can't make up my mind if the problem is with my Vigor 2800 or Asterisk 1.6.2. I know the Draytek is alleged to suffer from nat 'issues' but I did not have the issue with 1.6.1 - so I'm wondering if something has changed? The Draytek offers 'NAT & Routed' on a single device - so my Asterisk sits on a Public IP, and I have a

I have the OpenSSH regression tests hooked up to run in Debian and Ubuntu's "autopkgtest" system, so that they're automatically run on uploads of OpenSSH itself or any of its dependencies. This is especially good for enforcing interoperability between it and other SSH implementations, but it's also pretty good for throwing up occasional extremely-hard-to-debug failures since

Okay, a new method: Helpers#URL. Here''s the rundown. == Helpers#URL == Builds a complete URL to a controller route or a path, returning a URI object. Basically, an absolute URL which can replace the ugly `self / R(...)` in your controllers. Assuming the Hoodwink.d app is mounted at http://localhost:3301/hoodwinkd/, in a controller or view you''ll see the following

From a 'sip debug': Sip read: SIP/2.0 100 Trying From: "Evert"<sip:[username]@[my ext. IP]>;tag=as6e18534e To: <sip:[dialled number]@[SIP server of VoIP provider]> Call-ID: 6cbf41c25281f08b2e7bbc5043061975@[my ext. IP] CSeq: 102 INVITE Via:SIP/2.0/UDP [my ext. IP]:5060;branch=z9hG4bK4fd1045b Content-Length:0 7 headers, 0 lines Sip read: SIP/2.0 403 Forbidden

I have setup a trunk on Asterisk 11.7 to an Avaya Session Manager. Every time I try to send a call over it, the call gets rejected. Here is the sip debug trace. Could anyone tell me what may be going wrong? nxdasterisk-2*CLI> [Apr 23 08:20:59] WARNING[19047]: pbx_spool.c:309 safe_append: Unable to set utime on /var/spool/asterisk/outgoing/scott.call: Operation not permitted Audio is at 18380

Hi Damien, On Feb 23 10:28, Corinna Vinschen wrote: > On Feb 22 07:59, Damien Miller wrote: > > On Sat, 21 Feb 2015, Corinna Vinschen wrote: > > > - The failing last loop in the "forwarding" script as reported back > > > during 6.7 testing is still failing for me more often than not. It's > > > always the same reason, the script tries to use

CentOS Errata and Bugfix Advisory 2017:3301 Upstream details at : https://access.redhat.com/errata/RHBA-2017:3301 The following updated files have been uploaded and are currently syncing to the mirrors: ( sha256sum Filename ) x86_64: 228629701ba22f57356bc1f68ee3802df49d8f72f0b0a95b0ac68fda1a819dfd pki-base-10.4.1-17.el7_4.noarch.rpm

Hello, I can't find the information on the wiki :( When is the quota recalculated after a mail deletion ? For instance, I am running low of storage and I use Thunderbird to delete large mail. I only notice a recalculation when I quit Thunderbirdb and I relaunch it. Even, with doveadm CLI, as long as Thunderbird is not disconnected on the client side, the server didn't recalculate the

Hello, Suppose I have the dataset shown below. The amount of observations is too massive to get a nice geom_point and smoother on top. What I would like to do is to bin the data first. The data is indexed by Time (minutes from 1 to 120 i.e. two hours of System benchmarking). Option 1) group the data by Time i.e. minute 1, minute 2, etc and within each group create bins of N consecutive

Hi All, I have been learning some ruby and some Camping at the same time by implementing a little webapp to track expenses (one man''s blog...) that includes the ability to tag entries with 1 or more tags. I wanted to offer the possibility to narrow the view of expenses by adding tags to a filter (this works) and I also wanted this filter to be reflected in the URL. Like so: normal URL:

Hello all, I'm new to R and trying to figure out how to perform calculations on a large dataset (300 000 datapoints). I have already made some code to do this but it is awfully slow. What I want to do is add a new column for each "rep_ " column where I have taken each value and divide it by the mean of all values where "PlateNo" is the same. My data is in the following

Hi all! I'm trying to have asterisk route all outgoing calls out via my VOIP provider. exten => _NXXXXXXX,1,Dial,SIP/BYEXTENSION@VOIP seems to have them to in the direct direction. However, debug shows that my asterisk doesn't identify itself correctly: Sip read: SIP/2.0 100 Trying From: "Evert"<sip:asterisk@[my IP]>;tag=as0aca53fa To: <sip:[dialled