similar to: 1.2beta and PRI and CDR Corruption

I just released ruby-agi-1.1.0 Here is the release notes: addition of method 'jump_to' jump_to would take three arguments ex. jump_to(context, extension, priority) enhanced callerid, calleridname and calleridnumber. calleridnumber is an addtional method to this release. Regardless of Asterisk version above three caller methods would return uniform value. Such as,

Hi everybody, I have corrected this line in extensions.conf by stripping spaces off and now it executes: exten => 22999,1,VoiceMailMain(s${CALLERIDNUM}) when it runs, the mail box number is asked and password too. I expected no question were made, because I inserted CALLERIDNUMBER and s in front of box number. Anybody knows why? Thank to you all, very kind members of this list! Ciao Mauro

Hi, I've been trying to use the set_callerid function in the AGI. It sets the CallerIDname properly but I can't figure out how to set the CallerIDnumber. Is it at at possible ? Cheers. SL

hello, Need advise I've been attempting to get a winXP system added into the network of win95/98 systems and a Samba 2.2.7 server (redhat 7.3) current version of Samba version 2.2.7a-security-rollup-fix the win95/98 systems work fine. the winXP has trouble .. 1) It can not see the samba server OR other systems in the network neighborhoods. (it CAN ping the IPs tho .. ) I've read

hello I would like to join a Netapp fileserver to a Samba domain that uses an external Openldap server for authentication. Here's what I do and where it goes wrong (I've hidden the real ip addresses). I've also created a machine account for the Netapp server on the PDC.: What is the name of the Windows NT 4 domain? [WORKGROUP]: PCKLAS CIFS - Starting SMB protocol... Wed Jul 5

Release notes of ruby-agi-1.1.1 February 09, 2006 This is a bug fix release of ruby-agi. Below two bugs have fixed in this release. -- ReturnStatus#timeout? was not functional, which has fixed. -- AsteriskVariable#init_caller_variable updated to fix callerid bug which was returning 'nil' for number only callerid. method init_caller_variable is a private method that manage callerid,

Greetings, Legend: storage-gfs-3-prd - the first gluster. storage-1-saas - new gluster where "the first gluster" had to be migrated. storage-gfs-4-prd - the second gluster (which had to be migrated later). I've started command replace-brick: 'gluster volume replace-brick sa_bookshelf storage-gfs-3-prd:/ydp/shared storage-1-saas:/ydp/shared start' During that Virtual

I am working with a customer and their SIP provider is IPitimi. The customer needs to sometimes provide various caller id number for the calls going to IPitimi. They are processing calls for multiple businesses who want their caller id to show up. When no caller id is provided, the From must be the DID at ipitimi ip address and caller id is DID at customer IP address. When caller id is

Phone: Cisco 7960 I tried the following in my extensions.conf: exten => _*1XX,1,Voicemail(u${EXTEN:1}) exten => _*1XX,2,Hangup When I pick up the phone and dial *120 (my extension is 120) it doesn't go through, and says Reorder. For some reason, I don't think asterisk is picking up the asterisk digit. Anybody know how I can make this work using *<EXT>? Thanks.

Hello, I maintain audio processing part of code in sipXtapi -- open-source VoIP library. And when I engaged Speex library, I found that its Windows project files are not up to date and use strange settings. They miss several source files and use single-threaded static runtime libraries. I see no benefits of using static and moreover single-threaded libraries. In the modern world very small amount

Hi, alhtough i googled for details concerning ASTCC i did not found an aswer to the following: Should i put in my extensions.conf the configuration of the astcc? I ask this because as i see it, in the end of the extensions.conf there is an include statement : #include /var/lib/astcc/astcc-exten.conf Should the config been done in the astcc-exten.conf file or the initial extensions.conf

Hello, I've installed the latest samba (3.0.1) on my local test mashine and get the fullowing error while opening a file when I already have the status of: Samba version 3.0.1 PID Username Group Machine ------------------------------------------------------------------- 21236 alex users alex (192.168.0.111) Service pid machine

I have asterisk 1.2.18 running on a new system we just installed. Although I've used AGIs many times in the past, I'm stumped on this one. It may just be a simple issue that I need another eyeset to look at. My AGI does the following: #!/usr/bin/perl #Load a few modules... use Asterisk::AGI; use DBI; $AGI = new Asterisk::AGI; #Grab input from Asterisk my %input =

A full example with dovecot debug symbols activated (note since I have activate valgrind, no more segfault it?s normal I presume ): Dec 4 12:09:40 dovecot: auth-worker: Error: ==3071== Invalid read of size 8 Dec 4 12:09:40 dovecot: auth-worker: Error: ==3053== Invalid read of size 8 Dec 4 12:09:40 dovecot: auth-worker: Error: ==3057== Invalid read of size 8 Dec 4 12:09:40 dovecot:

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 am getting this on the console once people call in -- outgoing agentcall, to agent '1001', on 'Local/3044@local-fd6d,1' -- Called Agent/1001 -- Executing Macro("Local/3044@local-fd6d,2", "sipline|3044") in new stack -- Executing Dial("Local/3044@local-fd6d,2", "SIP/3044|20|t") in new stack -- Called 3044 --

The setup: Inc SIP Call -> Asterisk 1 -- IAX --> Asterisk 2 -- SIP --> phone (3044) Asterisk 1 shows the following: (1.0.3) -- Executing Goto("SIP/XX.XX.XX.XX-0819f590", "cytel-internal|3044|1") in new stack -- Goto (cytel-internal,3044,1) -- Executing Dial("SIP/XX.XX.XX.XX-0819f590",

Hello to all When my SNOM (300 or 320) loses Internet connectivity, it loses its Asterisk registration (ok, thats normal). But when the phone is back online, he doesn't try to register in Asterisk. I believe this happens to avoid flooding the private LANs when the Internet link is lost.... but the problem is that the phones don't try to re-register in the future.... Sometimes it stays

I'm not a PRI expert and therefore don't know what this debug stuff means for PRI, so if anyone can help me here... I'm running the latest libpri and zaptel from CVS. Keep in mind that everything works fine when using the STABLE libpri and zaptel. I am NOT running CVS asterisk. I am running 1.0.6. asterisk*CLI> pri intense debug span 1 Enabled EXTENSIVE debugging on span 1 T203

I have 2 asterisk boxes: asterisk-alpha (running 1.0.3) and dev-asterisk (running latest CVS). I am the only SIP user on dev, everyone else in the office is on alpha. If someone dials my extension, it should go IAX to the dev server and the dev server should ring me. Here is what I see on the dev machine's console: -- Accepting AUTHENTICATED call from 192.168.1.25, requested format = 256,