similar to: consultative transfer with zaptel

New application for asterisk : axra axra runs separately. developped in C++. it dialogs with asterisk through agi calls and through the manager api. it proccesses phone calls through the dial plan (agi) and concurently through the manager api. axra currently provides consultative transfer for SIP and IAX2 phones. this should easily be extended to any phone technology. hopefully, axra will

RFC 5229 gives the following example: if address :matches ["To", "Cc"] ["coyote@**.com", "wile@**.com"] { # ${0} is the matching address # ${1} is always the empty string # ${2} is part of the domain name ("ACME.Example") fileinto "INBOX.business.${2}"; stop; And I do not understand

Hi All, I download the app_asyncgoto.c, compile the app_asyncgoto.so. Then according to this page http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO ; when I dial ,there have this warning: -- Executing AsyncGoto("SIP/111-086497c8", "SIP/113-08674628|dynamic-nway|111|1") in new stack Feb 2 16:53:10 DEBUG[4218]: app_asyncgoto.c:95 asyncgoto_exec: Attempting

If I have in my dialplan [AgentQ] exten => _XX.,1,Dial(Sip/{$exten},120,g) exten => _XX.,2,NoOP(here we are) where [AgentQ] is called by the queue command to a member added by addqueuemember(Local/99@AgentQ) why don't I get to the NoOp if the agent hangs up during the announcement message (to the agent) ? I see in the app_dial.c program that the "g" flag is tested thus:

our * crashed twice in a month with segmentation fault & a core dump. here's the stack trace: #0 0xb7e11965 in mallopt () from /lib/tls/libc.so.6 #1 0xb7e10c43 in malloc () from /lib/tls/libc.so.6 #2 0xb7e17090 in strdup () from /lib/tls/libc.so.6 #3 0x08057ada in ast_verbose (fmt=0x0) at logger.c:879 #4 0xb7b29009 in moh_files_release (chan=0x9455ca0, data=0xb657be48) at

Hi, I try to set up attended transfer in my Asterisk Box . My features.conf look like this: [general] parkext => 100 parkpos => 1-5 context => parkedcalls parkingtime => 100 transferdigittimeout => 3l courtesytone = beep xfersound = beep xferfailsound = invalid featuredigittimeout = 500 ;adsipark = yes pickupexten = *8 [featuremap] atxfer => *2 blindxfer => # disconnect

We need to configure asterisk to authenticate two sip ATAs, but the stream must go directly from one to another ata without tuching asterisk. Is this possible adding canreinvite=yes into sip.conf? is it true laso if asterisk doesn't recognize the spd (t38)? thanks Rosario -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I'm using the following setup : Alice ---- IPPhone ------<LAN>----- Media gateway ----<PSTN> ------- Phone ---- Bob For certain calls, users complains about echo : they can ear their own voice in their handset, though media gateway echo cancel is turned on. I'm wondering how this echo cancelation engine is supposed to work. My understanding of echo is that most probably,

Hallo da mein English nicht so gut ist und bei der ?bersetzung auch noch Missverst?ndnisse auftretten k?nnten, hier das Orginal :-) Das Problem ist, dass bei der Kombination openssh pam und ldap, die Verbindung zum Ldapserver so lange offen gehalten wird bis die ssh Session geschlossen wird. Das ist nur bei SSH so ! Alle andere Dienste sprechen den Server an und schliessen nach Best?ttigung des

The following configuration has been working OK with Samba 3.0.1 Mandrake 9.0 PDC. Redhat 7.3 BDC. (both with ldapsam passdb backend). Solaris 8 Memberserver. After upgrade to 3.0.3pre2, exactly the same configuration - with exactly the same smb.conf - produces an error, when you try to add a user or group in the security settings of a file or directory through MS Explorer. The message text is

Hi, I am trying to achieve 3-way conferencing taking hint from wiki link http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO Here is the scenario: 1. user "ua1" calls user "ca1" 2. "ua1" then presses the feature code "*0" to redirect "ca1" to conference room 300 3. "ua1" then dials the user "33" 4. user

I download app_valetparking.c from http://www.loligo.com/asterisk/misc/apps/app_valetparking.c and followed the directions on http://www.loligo.com/asterisk/misc/apps/app_valetparking.README I am using asterisk-1.0.0 any suggestions [root@localhost asterisk]# astxs -install apps/app_valetparking.c gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarat ions -g -Iinclude

Here is the full "transaction" -- outgoing agentcall, to agent '1001', on 'Local/3044@local-4fee,1' -- Called Agent/1001 -- Executing Macro("Local/3044@local-4fee,2", "sipline|3044") in new stack -- Executing Dial("Local/3044@local-4fee,2", "SIP/3044|20|t") in new stack -- Called 3044 -- SIP/3044-ea92 is

I have tried for several hour to get a mysql db connection to work, but have failed so far: This is a cpanel server running Mysql 4.1.19, fcgi with apache. Here is the output from script/console Loading development environment. /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.14.2/lib/active_record/connection_adapters/abstract/connection_specification.rb:194:in

On 10/21/20 11:15 AM, @lbutlr wrote: > RFC 5229 gives the following example: > > if address :matches ["To", "Cc"] ["coyote@**.com", > "wile@**.com"] { > # ${0} is the matching address > # ${1} is always the empty string > # ${2} is part of the domain name ("ACME.Example") >

I've upgraded a couple servers to the latest icescast2 and ices2 sources. Everything works well with relaying. However, I am getting some reports that after about 50 minutes the stream on the relay machine dies and people are unable to connect again using Winamp-2.91. They have to close up and restart Winamp. I am thinking this may be a problem with the tree separate audio streams I am

hello all- Long time listener, first time caller. This is a great list and has given me tons of help as I've set up * for the first time. I've got an asterisk system up and running at a new company, and it does about 99% of what we need it to do. TelephonyWare has been our equipment supplier, and has been great with support, but I've got an issue that has us both stumped.

Sorry for the crosspost (this was also posted to asterisk-at-uc-dot-org) but I haven't got a response. I have a cell phone added to a queue as a local extension (member => Local/299). I want the cell phone to be able to reject calls to the queue without the person sitting in the queue being hung up on, etc. The way my dialplan is set up, the person hits 1 to answer the call and any other

Hi. Since a customer requested us that feature, I wrote this little patch for app_dial to allow to play an announcement to the called party, as soon he answers. you can define the file to play in the dial() option, using A(filename). for example: exten => blah,1,Dial(Zap/blah,30,rA(/my/own/announce)Tt) that doesn't break anything ... feel free to blame me for anything bad this patch

The Asterisk Development Team has announced the release of Asterisk 1.8.2. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.8.2 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this release: