similar to: Oh323 channel problem

Hi all, we use OpenSER together with Asterisk. All SIP users registers with OpenSER and asterisk is doing the voicemail thing. We use the Asterisk RealtimeArchitecture for voicemail users and SIP peers. The database table for the sip peers is a view from the OpenSER subscriber table. The MWI for a user will only work, if the user object (sip peer) is loaded into memory and visible with the CLI

Hello Users, My Setup is like this openser --Registrar asterisk --Callflow using asterisk-b2bua + radius for accounting My Intention was to generate a Acct-Stop Packet when there is a failure of RTP media from one of the UAC's( callee or caller) who is in dialog. so that the Caller will not be charged for Meaning less network problems Because there is no way asterisk knows about

https://bugzilla.netfilter.org/show_bug.cgi?id=1501 --- Comment #8 from marco.drummer at outlook.com --- (In reply to Phil Sutter from comment #7) I am currently using iptables v1.8.7 (nf_tables) on Ubuntu 22.04.2 LTS Almost all of my rules are converted to nft to make use of the advantages and simplifications in syntax. However since shifted port ranges are still not available I still have a

https://bugzilla.netfilter.org/show_bug.cgi?id=1501 --- Comment #9 from Phil Sutter <phil at nwl.cc> --- (In reply to marco.drummer from comment #8) > (In reply to Phil Sutter from comment #7) > > I am currently using iptables v1.8.7 (nf_tables) on Ubuntu 22.04.2 LTS > > Almost all of my rules are converted to nft to make use of the advantages > and simplifications in

Hi! I've read in the asterisk docs (AstARA.html) that realtime users/peers can't be notified (MWI with SIP NOTIFY) when they have new voicemail messages, because their object are not persistent in memory?! But that's what we really need!!! Is their any work in progress to get these things working? Or are their any known workarounds? I'm using asterisk version 1.2.1 with OBDC

Hi, does anybody know if asterisk can authenticate on a NOTIFY send to a peer. I use OpenSER as SIP-Proxy and asterisk as voicemail system with ODBC Support for voicemail-messages, voicemail-users and sip-peers/users. My SIP-Users register with OpenSER. Asterisk has two views on the OpenSER database for voicemail-users and sip-peers. Call-Routing, leaving voicemail-messages in database an MWI

Does anyone know how to do this? Just curious, ie SIP callflow A -- Asterisk -- B, RTP goes directly from A to B .. Matt

Hi all, i have the following setup PSTN -> routing server (asterisk 1.6.2.11) -> IAX -> callcenter asterisk 1.6.2.9 -> SIP -> agent Does work quit fine - then agent does have the abibility to transfer a call to a third party - the agent can initiate the transfer over a web interface - it does generate a asterisk manager atxfer request... So agent does initiate transfer - call

Hi All, I implemented a standalone ruby program that helps to my RoR app to execute processes in background, the web app adds a request to the ruby program, then this spawns a thread and the web app monitors the status until the process is completed. All things work perfect and nice. But, I noted that I''m getting multiple DB idle connections, I think (not sure) this behavior is OK

Hi all i have been succesfully using OpenH323 (oh323) for a few months. the versions are: asterisk CVS HEAD 19-07-2005, OpenH323 v1.13.5, PWlib v1.6.6, asterisk-oh323-0.7.2-pre1 I now have moved to Asterisk 1.2.4, so as per the directions i am using: Asterisk 1.2.4 pwlib_Mimas_patch2 openh323_Mimas_rc2 asterisk-oh323-0.7.3 The problem is that when asterisk starts it fails on loading the module

I know asterisk isn't a real SIP proxy and is more of a multi-protocol pbx with limited SIP support, but... ... is it possible if you have a central registration server that handles all of your dialplan routing and several asterisk PSTN gateways that it routes calls to for an outbound SIP conversation using reinvites and NOT have the registrar box try and send ANY RTP traffic back to the

SafiServer and SafiWorkshop 1.2 is here! This is a seminal release for us as the product is now more stable, powerful, and easy to use than ever. We've also added a new ActionStep "CallWSByWSDL" that allows you to easily consume Web Services from your Saflet, providing you with even more integration possibilities for your IVR/Callflow applications. The release of this ActionStep

Hello I've Compiled openssh 3.9.p1 on AIX 5.2: <source Dir>/contrib/findssl.sh Searching for OpenSSL header files. 0x009060dfL /opt/freeware/include/openssl/opensslv.h Searching for OpenSSL shared library files. Searching for OpenSSL static library files. 0x009060dfL /opt/freeware/lib/libcrypto.a 0x009060dfL /opt/freeware/64/lib/libcrypto.a 0x009060dfL /usr/local/lib/libcrypto.a

When a user dials 999 he is always asked for the mailbox and has to enter his mailbox number and password. As I understand this shouldn't happen because the CALLERIDNUM is passed over to VoicemailMain. It's annoying to have to enter the number everytime ... The voice mail configuration is read from MySQL. We are using the CVS version from a few days ago. Extract from extensions.conf:

Hello everybody For network simulation purposes I am trying to run a Linux image with a PAX enabled grsec kernel on a Gentoo xen-3.1.1 with HVM. While the image boots flawlessly on real hardware the kernel does not really like the fully virtualized Xen/Qemu environment. It does not succeed to boot (for dmesg see attachment). I first tried with the grsec- patched 2.6.14.6 sources but it

Hi Does anybody know how to build the LibIAX2 from Virbiage? It has some nice features when using Firefly (Messaging, Status Indication). The source can be downloaded here: http://www.virbiage.com/3rdparty/. It does not contain any directions how to compile. Any hints? Thanks! Reto

Hi Does anybody if the X100P works in Switzerland? We can't get a line to PSTN. When I run zttool it shows me always a red alert. I can make and receive calls with an anlog phone plugged in the phone connector. I've compiled and configured the card according to the wiki. Everything seemed to be ok. Is there a way to debug this? Regards Reto

excuse for to find a desperate solution... but i'm boried to spent hours... ;) not in asterisk !!! ;) i use the festial-script.pl of Donny Kavanagh... but i want to change the language that festival uses, depending on a variable for the callerid. English/Spanish How i can tell the script the correct voice that festival needs to use?? like --language spanish --language english ...in normal

Hi everybody, I would like to create SIP call flow Diagram under Windows. Is anybody know a program to perform it? I have already Ethereal and I would like an explicit diagram just to show where something have problems... Thanks Ignace

Hi there Is there an easy function available in R for calculating the running mean? Reto -.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.- r-help mailing list -- Read http://www.ci.tuwien.ac.at/~hornik/R/R-FAQ.html Send "info", "help", or "[un]subscribe" (in the "body", not the subject !) To: r-help-request at