similar to: Are codec "capabilities bitmasks" different in IAX and SIP?

I am having a problem with an Asterisk server, in that when it is receiving a call from another Asterisk server using an IAX2 trunk the phone rings for 10 ms and then there is a hungup from asterisk and then the phone rings again before another hangup. The funny thing is that after I really hang up on the calling phone it repeats this as if I am still trying to call. Any Ideas?

Hi all, I've scoured the web, faq and mailing list archives to try and find an answer to this problem I'm having, but to no avail.... I'm trying to rsync a Cyrus IMAP message store "structure" from one system to another. (I say "structure" because I don't want to copy over the actual messages in the message store.) For those of you unfamiliar with the

Hello I have created an iax exten in my iax.conf file: [300] type=friend username=300 secret=*** context=default host=dynamic callerid="some name" <300> auth=md5 Then in my extensions.conf I have: exten => 300,1,Dial(IAX/${EXTEN},20) exten => 300,2,Hangup I can dial from iaxComm (a soft IAX client) and that works fine. But when I try to dial 300 get: WARNING[22077]:

I think I have it configured properly. ztcfg -vv shows it as channel 1 and zttool shows it as OK. But I can't dial out. When I try, it shows it arrive in teh right stack, but then issues the following errors: channel.c:1676 ast_request: No channel type registered for '{PSTN-1}' app_dial.c:554 dial_exec: Unable to create channel of type '{PSTN-1}' = = Everyone is busy at

There are two 64-bit bitmasks maintained in AttributeImpl.h<https://sdocc.itg.ti.com/ui#file:review=11893/version=393846>: - AvailableFunctionAttrs is part of the AttributeListImpl class, and - AvailableAttrs is part of the AttributeSetNode class Both of these assume that the number of available enum attributes is limited to 64. In fact, a static_assert in

I have spent the morning on this project, still without success. Summary: Yesterday I inadvertently unplugged my Grandstream phone. I might add I did a rebuild of my s/w from CVS at the same time. Since then, the Budgetone seems to talk SIP just fine, but the RTP being sent to it by asterisk "doesn't make any sound." It was suggested I do a factory reset of the phone, which I

Hi Folks, I'm running Debian-4.0 Etch, installed last September from a DVD, and regularly updated as things arise. I have R version 2.4.0 Patched (2006-11-25 r39997) installed (initially at the time of first installation of Debian, as provided by Debian), along with a variety of packages. I'd like to be able to connect to the CRAN repositories for Debian R, for updates etc. When I visit

Hello, I'm planning to add Asterisk as VoIP gateway between two existing offices, A and B. A has a DCS Compact II PBX with a spare 4BRI card which I plan to connect to an ISDN card, probably an EICON Diva Server 4BRI) on my VoIP gateway. The gateway will be connected via IP to the B office's gateway. B has a DCS Compact PBX without spare lines. To handle this, I thought I

(Server :Mandrake 7, Samba 2.0.6) (Client : PC Win95 and Win 98) Samba's work fine since one year but yesterday We had a big problem in my college Lots of PC's client can't connect (invalid password). In the samba's log we read : > [2000/05/19 12:15:21, 1] smbd/service.c:close_cnum(568) > renoir (192.168.84.189) closed connection to service valancee > [2000/05/19

Hello all I need your support in a big decision in front of two alternatives related with *. I must buy an E1, in order to manage 30 channels, given this big price; or I could opt for a 15 BRIs without cost to replace the same number of channels, and the question that inmediatly emerge is : ? can asterisk manage 15 BRIs ? If yes to the latter, could posible somo guide, for instance, wich Digium

Hello, I am getting a GPG error with the ubuntu repository at the bristol UK mirror. When my source.list has this line: deb http://www.stats.bris.ac.uk/R/bin/linux/ubuntu/ intrepid/ On an "apt-get update" you get this: W: GPG error: http://www.stats.bris.ac.uk intrepid/ Release: The following signatures were invalid: BADSIG D67FC6EAE2A11821 Vincent Goulet <vincent.goulet at

Hi, $ more /etc/apt/sources.list.d/additional-repositories.list deb http://www.stats.bris.ac.uk/R/bin/linux/debian wheezy-cran3/ deb-src http://www.stats.bris.ac.uk/R/bin/linux/debian wheezy-cran3/ $ sudo apt-get update $ sudo apt-get install r-cran-rgl # works great $ sudo apt-get build-dep r-cran-rgl Reading package lists... Done Building dependency tree Reading state information...

Sorry for posting this again, but it seems to have become attached to another thread. Guess I replied to another message instead of starting a new one... Hi, I'm trying to setup a call forwarding rule so that when an extention doesn't answer the call is forwarded to my mobile. I'm using voiptalk.org for incoming and outgoing calls and SIP phones for extentions (so all IP based -

Hi all, I am very new to R and this might be a simple question but I have looked everywhere you suggest before writing to you. I am trying to change font type from san-serif to a serif (Times New Romans) on all labels and axis of my boxplot. I have used this function in other plots before, e.g.: plot(residuals~lnlifespan, data=mydata, pch=psymb, font=6, xlab="ln reproductive

I am seeing some curious behaviour with a 1.2.32 system, which I do not understand and so can't work out how to fix it. I have a PRI routed to context default. Here is the complete default context: [default] exten => _9X.,1,Dial(IAX2/m1peer/${EXTEN:1}) exten => _20XX,1,Dial(IAX2/sipeer/${EXTEN}) exten => _X.,1,Dial(IAX2/m1peer/${EXTEN}) exten =>

Hi, I have the following problem that when asterisk receives SIP response 302 it cannot forward the call I get such debug: [Jan 4 10:43:27] WARNING[18671]: channel.c:3281 ast_request: No channel type registered for 'Local' [Jan 4 10:43:27] NOTICE[18671]: app_dial.c:505 wait_for_answer: Unable to create local channel for call forward to 'Local/poczta at routing-sip' (cause = 66)

Hi All - Well, after happily existing in a one office environment with asterisk for a few months, I've now decided to start adding in our other offices with their own * boxes and IAX connections (over VPN). Unfortunately, I'm an idiot and I can't get it to work. I'm having some kind of problem with codecs, I guess, but I don't understand what or why. When trying to use

I'm trying to produce some enhancements to one of the applications, and am trying to use ast_request("zap", format, "pseudo") to create a new channel on /dev/zap/pseudo, which I can then bind to a zaptel conference and play a stream to it. I've been using as inspiration the Radio Repeater app, app_rpt.c, which uses this technique to play idents and announcements.

Hi everyone, I just got myself a IAXy device and am trying to integrate it to our asterisk server. I configured the IAXy and it is registering and I get a dial-tone. If I try calling another SIP device, and I get "can't translate IAX2 to SIP" How can I make my IAX device communicate with a SIP device (and vice-versa)? Here's what the log says: -- Executing

Hi My hardware PBX run asterisk on vxworks,Because the vxworks not support perl. Now I want to add a callback function to my pbx. now it can store Caller and Called party numbers in queue when Called party is busy Then I malloc a new ast_channel to call.It is should use ast_get_channel_by_exten_locked() or ast_channel_alloc() , my program as follow,But it isn't work, anyone know how to