similar to: JABBER_STATUS CODE 7

Lordy, I've been having problems with this darn thing, so I hope someone can help me. :s My troubles started when I downloaded the latest erlang and ejabberd packages. I crashed and burned very quickly, trying two or three different versions of erlang along with several of ejabberd 2.0.x. Finally, after a week of pain, I admitted defeat, wiped the whole lot and installed the binary on the

Hi! I'm trying to install an ejabberd server on my network. Ejabberd supports LDAP - but when I tried to get it to work, it opened a whole can of worms. Basically where I'm at now, if I use ldapseaerch to try and search for, say, a username - I get NT_STATUS_LOGON_FAILURE - so my question is, does samba have an ldap server build in, or do I need OpenLDAP?

Hi all, For one of my inverstigations it looks like i'm back to "square one" I'm trying to accept an incoming xmpp call and forward it conditionally to a sip, isdn, or voicemail. No google is involved as i use a local xmpp server (ejabberd) I was experimenting on 1.8.15.1 (with jabber.conf, jingle.conf), but some suggested me to have a look at asterisk11,so i did... I

Hi list, I have a big doubt!, I have some users with ejabberd and am using motif to make some calls to extensions, here works fine, the problem is when I want to send a message to another user on ejabberd and asterisk take this message as part him, like a sip message , the other user does not receive this message xmpp User A xmpp == Chat to == User B xmpp (not receive the message) look cli

Good day to all! We would like to authenticate our ejabberd against Dovecot with perl there's such a script http://www.ejabberd.im/files/contributions/check_dovecot.pl.txt that script uses following perl library - http://cpansearch.perl.org/src/SASHA/Authen-SASL-Authd-0.04/lib/Authen/SASL/Authd.pm Authentication is OK, but sub user_dovecot wich queues information about a specific user

Hi all, What other open source tools are people using for this? I was looking at Openfire and their asterisk plugin. Is it easy to roll your own with res_jabber.so ?? Thanks. -- Sent from my mobile device http://www.suretecsystems.com/services/openldap/

Hello Everyone, I am using this bash script to pull resource id http://fpaste.org/84173/51169913/ this whole macro http://fpaste.org/84174/51176013/ that what I see when dialplan ran Executing [s at macro-missed-call-in:3] Set("SIP/babytel-00000022", "RES=9c32ecc4 -- ") in new stack -- Executing [s at macro-missed-call-in:4] GotoIf("SIP/babytel-00000022",

Hi, I was woredering if you could help me to put into practice this solution. The idea: Create a IVR-Voicemail The scene: PSTN------/6------PBX--------/12--------- Internos | /4 ports | IVR-Voicemail The Operation: 1)Where a call enters from the PSTN, the PBX flashes and

Hi All I am developing a client that uses libjingle to do xmpp stuff with ejabberd. I can also make audio calls between those clients. What I am trying to archive now is to send calls to pstn using jingle. I was told in the jingle-dev community that asterisk can do that. Is there any way to send jingle audio calls to asterisk and will it understand them ? If yes..can I forward those calls to PSTN

Given the recent announcement about Google slimming their support for public interconnection with XMPP, can anybody comment on where this leaves the XMPP support in Asterisk? In particular, I notice many of the references to XMPP on the wiki link to https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google which seems to suggest that XMPP support and Google Talk support are one and the

That little Digium adapter S101I is a nice and compact (good for travel) but has a serious limitation, it doesn't support bridging. If I have four/five of these units registered to my asterisk server (over the Internet) and with standard DSL or Cable connection all for of them connected and in use (utilizing G711); my upload bandwidth will get saturated as all of them have to go via my

Hi list, I write on the list looking for help, buy a openvox gw gsm for four channels and I'm a little disappointed with the support openvox, for some reason , The call doesn?t get trough support tells me it was my asterisk server, but does not really work me and my internal calls are working perfectly, I tested with another sangoma FXO gateway and works perfectly. the problem is that

I'm working on calls coming in to an asterisk box as H.323, and going out as SIP to a remote device (a VoiceMaster). The remote device is refusing the calls with SIP error 406 (Not Acceptable). I have attached the SIP debug output below. It looks like codecs overlaps - can anyone see why the call is being refused? (Note that I'm not registering with the remote SIP device, just

I don't mean to be a troll in any way shape or form. I was on IRC last night and I observed the following convo. below. What do you guys make of it ? [02:14] <bkw__> Let me tell you how chidlish digium and Mark Spencer is. I walk into a restaurant with them all here at Astricon wearing my sangoma shirt and he asked me to leave. [02:15] <Dovid> u serious ? [02:15] *** mog

Hi list, I need your help ,I have an incoming call x the ivr and the operator takes the call. ext "101" , If a second call reenters and the operator is talking, I want to send to the extension 102 I use the Variable DIALSTATUS , but not working check IVR [IVRINMA] exten => s,1,Wait(1) exten => s,n,Set(CHANNEL(language)=es) same=> n,Set(TIMEOUT(digit)=4) same=>

Excuse my ignorance (I just got crap on the #centos IRC channel for this question), but is there a (easy!) way to have and IRC and/or Jabber server relay a login to a Microsoft Active Directory server for authentication? If there's a better question to ask this question, please point me in that direction, and I'll be happy to do so Thanks -------------- next part -------------- An HTML

Processing commands for control at bugs.debian.org: > ## dear clint. > ## > ## release goals are release goals and not release blockers. > ## please learn the difference and discuss this beforehand. > ## > ## thanks. > ## > > severity 382189 serious > ## > Bug #382189 [nbd-server] no IPv6 support > ## > Severity set to 'serious' from

Processing commands for control at bugs.debian.org: > severity 382189 serious Bug #382189 [nbd-server] no IPv6 support Severity set to 'serious' from 'important' > severity 384372 serious Bug #384372 [libadns1] libadns1: Cannot query IPv6 DNS servers Severity set to 'serious' from 'important' > severity 517299 serious Bug #517299 [klibido] klibido: needs

You could use MTR command. Its a trace route improved. Marlon Araujo > On Jan 20, 2015, at 08:59, asterisk-users-request at lists.digium.com wrote: > > Send asterisk-users mailing list submissions to > asterisk-users at lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.digium.com/mailman/listinfo/asterisk-users > or,

Senhores algu?m pode me passar os valores aproximados para realizar essas configura??es, uma amigo n?o sabe como e quanto cobrar para fazer as configura??es para um cliente e fica me perguntando, eu cobro por hora, e o senhores ? O servi?os s?o: - Configura??o do PABX digital - Configura??o de URA - Configura??o de redirecionamento para celular por operadora ATT -- Cl?udio Duarte (61)