similar to: Receiving and processing unsolicited XMPP messages with Asterisk 11

I'm trying to get my Asterisk 11 test box set up with XMPP, having troubles with JabberSend(). My jabber.conf file is as follows: [general] debug=no autoprune=no [testaccount] type=client serverhost=my.jabber.server username=myaccount at my.jabber.server secret=mypassword port=jabberport usetls=yes usesasl=yes xmpp show connections gives the following output from the console:

I've been working on an interactive XMPP interface so users at my office can interact with the timeclock and queues by XMPP (in addition to IVR menu, which has been running just fine for quite a while before the XMPP interface). I'm using sendtodialplan=yes to handling the incoming unsolicited messages, and typically will have at least one point of interaction where Asterisk requests

Hi all,I hope to find a solution with the help of the list, I'm trying to get the status of my extensions with ejabberd , the idea is to visualize my users ejabberd incoming calls or missed. I'm testing with my operator extension with this code but only get the missed call notification does not show me where the call is coming. my piece of code [operadora] exten =>

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

We have a recently deployed server with a new TDM2400 card that will not put dialtone or audio on FXS ports after the physical server restarts (though they will ring if called, there's just no audio on the line if the phone at the other end picks up). The symptom can be resolved by stopping Asterisk, restarting DAHDI, and then restarting Asterisk. So far, this has happened on both times the

2015-03-18 11:13 GMT-06:00 ricky gutierrez <xserverlinux at gmail.com>: > Hi , I'm trying to apply this patch from the source asterisk > asterisk-11.16.0 and when I apply it shows me this message > > asterisk-11.16.0]#patch -p0 < refs > patch: **** Only garbage was found in the patch input. > > is the correct way to apply the patch or am I doing wrong? >

I am having difficulties with customer-bound DTMF being very short & clipped off (and basically unusable, as systems on the customer side aren't recognizing the DTMF digits, and I can barely tell that DTMF is there when I listen on a handset). My system set up as follows: PSTN <--> Metaswitch <-SIP-> Asterisk <-SIP or IAX2-> CPE Asterisk is running Asterisk 10.4.0 on a

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 , this is a bug? ERROR[361]: res_xmpp.c:3468 xmpp_client_reconnect: No XMPP connection available when trying to connect client regardss -- rickygm http://gnuforever.homelinux.com

I run a hotdesking system based on the example from Asterisk: The Definitive Guide. Calls come into the [hotdesk] context, which verifies the phone has a logged in user and sends the call to users,${EXTEN},1 if there is a user logged in. The [users] context then includes several other contexts for internal/external call handling, as follows: [users] include => internal include =>

Hello, our gateway receive incoming calls from an outside gateway for multiple DIDs. For some of them we want fax detection, for other no. To do so, faxdetect is set to yes, but how to disable the fax detection for a specific incoming DID? For those DIDs, we just want to forward the call to a real fax machine DID which will do the job. Thanks for any hint Regards -- Daniel

I've been tasked with finding and implementing a CDR/Queue analyzer to provide information to management about the call center's performance. My Google-fu seems to be returning a lot of things that are more or less abandoned projects. Does anyone have any recommended solutions for a CentOS 6 / Asterisk 10 "vanilla" server? Not opposed to something commercial, provided it

In testing Asterisk 11, I've found that Asterisk doesn't seem to be sending BLF updates to SIP peers that have subscribed to a hint looking at a Custom device if that Custom device state is RINGING or RING_INUSE. All other states seem to be working correctly. The hint section of the dialplan is: [hints] exten => _3XX,hint,Custom:${EXTEN} Console shows the following for core show

I have been working with distributed device states in Asterisk using XMPP attached to an OpenFire server. I have it working well across two servers and want to roll it out across every server in my company. All servers are Asterisk 11.6.0. I am running into a problem that seems like it should be a bit easier to solve than it is seeming to be. On the third server I am rolling into this

Is there a way I can change the ring cadence on FXS lines on a system using a Digium Wildcard TDM2400 card? I recently deployed a new phone system, and the customer has a few POTS phones in areas where they don't have data network services, so we're using the FXS lines to provide dialtone at those outbuildings. The old phone system would ring those phones with a short ring-short

Before I go down a rabbit hole, does the mwi publish/subscription work for non SIP phones? For instance, I have a single voicemail server, connected to multiple asterisk boxes via SIP. On each of those servers, there are a mix of SIP and SCCP phones attached. Currently, I'm using res_xmpp to distribute mwi from the voicemail server to the endpoint servers. Would this type of setup work

I've been spending the day trying to get IMAP_STORAGE on my test box, to evaluate for production, but I'm having no luck getting uw-imap to build. I've tried installing it from an upstream package, but Asterisk still isn't finding it to compile -with-imap. My google searches have turned up very little for documentation on dependencies, gotchas, etc for either item, so I'm

Hi everybody, I've a matter with the queue annoucement with the "thereare", because if I put just one member in my configuration (member => SIP/2098), the ivr gave me that I was the firt or second in the next at the queue. But the problem is, if I add one member (eg: member => SIP/2098 and member => SIP/2099), the ivr don't gave me the range but It play the

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

I'm running asterisk 16 on Fedora 31. If I start asterisk as user asterisk, all goes well. But if I start asterisk from systemd: asterisk[1411]: [Jan 21 19:36:47] ERROR[1411]: res_sorcery_config.c:320 sorcery_config_internal_load: Unable to load config file 'pjsip.conf' Jan 21 19:36:47 asterisk.riverside asterisk[1411]: [Jan 21 19:36:47] ERROR[1411]: config_options.c:710