similar to: Asterisk 1.8 Realtime Queue not working

I have asterisk 1.8.0 installed and I am not able to forward a voicemail from one users mailbox to another user. I have the user log into their mailbox press 8 to forward a message enter the extension of the user I wish to forward too I don't prepend a audio message and press # to send the message to the other user from a debug perspective I don't see any errors. The only message I see

I am trying to set up Hunt Groups and I am having some issues. Here is what I am trying to do. All my users actually register with OpenSIPS. Asterisk is using Realtime and I have set up a MySQL View Table so that Asterisk see's all the SIP users info that OpenSIPS has. This is what I have configured queues.conf ---------------------------------- [irock.com] strategy=leastrecent

Hello, I have a partially working Realtime Queue using unixODBC and Firebird. I followed the steps on asterisk-guru and the voip-info asterisk message boards but have been having no success. I say that my queue is partially working because callers can join the queue. However the calls do not go to the agents even if the agent is not tied up and is available for calls. So the caller is stuck

Hi, ? I am trying to store queues.conf to a MySQL database using dynamic realtime. I have a working ODBC connection and the queueing system already works but I want to store the queues.conf file to a database. I am following the guide from Asterisk the definitive guide, the ebook can be found at: http://ofps.oreilly.com/titles/9781449332426/asterisk-DB.html ? I have a database called asterisk

Hello, I'm using MySQL realtime Call Queues (table /queues/ and table /queue_members/). I would like to ring the members of the call queue in a certain order. Therefore I use ring strategy /lineair /and I put the members into the table /queue_members/ in the order in which they have to be rang. So I have the queue : | name | musicclass | announce | context | timeout |

Hi all, I'm trying to migrate messages from icewarp (merak) mailserver to dovecot via dsync, IMAP migration is looking fine, but I'm unable to migrate pop3 uidls from originating server, probably is something wrong with configuration, but I don't know what. The only documentation i found is on the wiki: http://wiki2.dovecot.org/Migration/Dsync where is bad writen mail_plugins =

Hi... I just had a real quick and simple question... I have a asterisk implementation setup w/ real time off of a mySQL database for SIP peers and queues, voicemail, agents etc... I after the upgrade to asterisk 1.4.3 there are some new configuration features i would like to use. I was wondering if i could just add to the database table a column for the new config option? if this will work or

Queue not sending call to Agent I am having an issue and i am not sure if it is a bug or a config issue. I was originally running Asterisk 1.8.1.1 when I noticed this issue. I upgraded to 1.8.4.2 to see if that would fix it but it didn't. The issue is that I have a call queue and the agent dials a number to log into the queue. When someone calls the queue the first time the call is

Hi, I am new to Asterisk. I'm using it behind a kamailio sip-router to provide voicemail boxes to sip-users. I followed these instruction: http://www.voip-info.org/wiki/view/Realtime+Integration+Of+Asterisk+1.4+With+Kamailio+1.5.x to set everything up, using ARA with a MySQL DB. After a few tweaks "everything" is basically working, however, a few questions remain that I could not

Hi again I'm trying to authenticate a user against an LDAP Server (well, our AD, but it can LDAP). This is my configuration: hosts = my.server.local auth_bind = yes ldap_version = 3 base = CN=Person,CN=Schema,CN=Configuration,DC=company,DC=local scope = subtree user_attrs = \ =home=/home/imapproxy/%u, \ =mail=maildir:/home/imapproxy/%u pass_attrs = uid=%u, userPassword=%w

I have been successfully connected (incoming and outgoing) to FWD for a very long time. A few months ago, I changed from SIP-based FWD service to IAX2-based, and that went fine as well, both incoming and outgoing. At the time, I was running Asterisk 1.0.3 Stable. I rarely use the service, so other than noticing that I was always successfully registered to FWD, I didn't make or receive calls

With 1.6.1 svn: [2009-04-26 15:01:00] NOTICE[1844]: chan_sip.c:9927 sip_reg_timeout: -- Registration for '17470121145 at proxy01.sipphone.com' timed out, trying again (Attempt #30) [2009-04-26 15:01:00] WARNING[1844]: acl.c:376 ast_get_ip_or_srv: Unable to lookup 'proxy01.sipphone.com' [2009-04-26 15:01:00] WARNING[1844]: chan_sip.c:10037 transmit_register: Probably a DNS

Hello, Raw stats: Version:1.8.3.2 OS:Centos 5.6 Special setup: postgre database I am having a few queue issues with Asterisk specifically relating to breaking out from queues while on hold. The intent is that while someone is on hold they can press a key (lets say *) to break from the queue and go elsewhere (in this case to leave a message). However In all of my testing I am unable to get

Does anyone else have DTMF issues with SIPPhone? When calling into my DID, and entering, say, 1002. Sometimes it will recognize it properly (rarely), other times it will receive something different. Such as, 1102 or 1000, etc. Has anyone else been having these issues? I'm only accepting ulaw and alaw, and my relevant sip.conf information follows: [sipphone] type=peer

I've beaten myself bloody dealing with this one... No luck so far. In summary, incoming calls from Gizmo establish, but neither get nor send sound. Outbound calls to Gizmo work fine (well a bit choppy but work) My thought is that the SIP connection is being made fine, but the RTP is getting stopped / blocked / misdone somewhere. Here is the thing: Asterisk 2.5 on Linux (No hardware

for inbound calls, i can register context = from-sipphone register => 1747xxxxxxx:passwd@proxy01.sipphone.com but how do i configure to make outbound calls to them? exten => _1747XXXXXXX,1,GoTo(dial-sipphone,${EXTEN},1) .... [dial-sipphone] ; ; SIP to sipphone.com ; exten => _X.,1,Dial(SIP/${EXTEN}@??????) ^^^^^^

I have a sipphone.com account, with asterisk set to answer incoming calls, using the following settings (phone number and password omitted) in the Peer Details for the SIP Trunk: allow=ulaw context=from-pstn dtmfmode=rfc2833 fromdomain=proxy01.sipphone.com fromuser=1747xxxxxxx host=proxy01.sipphone.com insecure=very secret=xxxxx type=peer username=1747xxxxxxx The Asterisk machine is

I'll admit it. I'm a asterisk newbie (but no stranger to telephony). The setup is simple: two Grandstream BudgeTel 100 phones (SIPPhone specials) on a private segment calling to a Linux box acting as the segment's firewall with a leg on our public network. The phones are setup as SIP/phone1 (x1000) and SIP/phone2 (x1001), respectively (thanks to the Asterisk HOWTO). Getting IAX

I have the following in sip.conf. It was my understanding that this configuration (ie with deny/permit) would only allow connections from hosts 192.168.10.4 and 192.168.10.5. That doesn't seem to be the case. Asterisk is accepting INVITE's from other addresses. [a00090101] type=friend context=Company1 username=a00090101 ;secret=180 ;insecure=very host=dynamic mailbox=company1@vmusers

Hi list! I'm always trying to configure Dovecot to ask our LDAP-Server (AD) in order to authenticate the users. I really don'know what can I do wrong... I configured my Dovecot so: hosts = chimaera.company.local dn = CN=mailproxy,CN=Users,DC=company,DC=local dnpass = SECRET sasl_bind = no tls = no debug_level = -1 auth_bind = yes ldap_version = 3 base = dc=company,dc=local deref =