similar to: Asterisk 12 AMI Hold Event

I have this in sip show history for a particular channel marked as dead (should be removed) in sip show channels: 1. TxReqRel INVITE / 102 INVITE 2. Rx SIP/2.0 / 102 INVITE 3. CancelDestroy 4. Rx SIP/2.0 / 102 INVITE 5. CancelDestroy 6. Unhold SIP/2.0 7. Rx SIP/2.0 / 102 INVITE 8. CancelDestroy 9. Unhold SIP/2.0 10. Rx

Hello, I think I met a case similar to the one solved by [1] . Quoting this case : * res_pjsip: Handle deferred SDP hold/unhold properly. Some SIP devices indicate hold/unhold using deferred SDP reinvites. In other words, they provide no SDP in the reinvite. A typical transaction that starts hold might look something like this: * Device sends reinvite with no SDP * Asterisk

Through the AMI how can I tell if a call is on hold or not? I am using 1.4.X Thanks, jerry

I am implemented a simple project management application. Each project has various events, and each event can be a different type. Some event information is common, such as name, start date, close date, and comments. I have a projects table which has_many events. My plan is to have several sub-event tables, like event_get_access which will contain an event_id field to link it to table events as

I got a very strange problem with call-hold function. For calls that come in from PSTN and route to a SIP extension. If I put the call on hold, I cannot unhold the call after. The caller would be left with hold music forever. A warning message would be shown on the console usually a few seconds after putting the call on hold: WARNING[17428]: chan_sip.c:686 retrans_pkt: Maximum retries

Dear list, this one is quite strange. It has happened on my own server a few days ago that the smbd process terminates silently when I write a file to the samba server. It never happens when reading, it is completely independent from the amount of traffic and it occurs several times a day in odd intervals. I ran smbd with a log level of 3, just to see what I already have seen in syslog (level 1):

Using X-Lite FWD soft phone, I can register, get to the 'demo' menu extension, but that's it. Audio starts, then after a few seconds stops, with packets still being passed. Anyoen have any clues? Yes there are firewalls between here and there, yes there is NAT at my end...What ports need punching, is rfc2833 the correct settign or should I use inband or info? TIA, I just

Hi, a few days ago, we switched our site from pop3 to imap. We are running about 20 users with Netscape 7.0 on a FreeBSD-4.8 Server . I installed dovecot 0.9.99.1 via the freebsd-ports collection, which was as easy as make install and start dovecot. Our users mailboxes are still standart mbox format in /var/mail/(user), as it is default in Freebsd. Now we suffer from a realy ugly problem,

Hi, I have the following symptoms: Call X-lite / Nokia E51 X-lite press hold: Nokia DOES hear MOH Nokia press hold: X-lite does NOT hear MOH Call X-lite / SCCP phone MOH works as supposed Call SCCP phone / Nokia E51 SCCP press hold: Nokia DOES hear MOH Nokia press hold: X-lite does NOT hear MOH In addition, the BLF on the SCCP phones does NOT show the hinted SIP extension on hold. With 1.4

Thanks Buddy, I got it working. It was giving error when running make 'undefined reference to `request_init' etc' then I ran it with gmake LDFLAGS+=-lwrap and it worked and everything is working fine. Thanks again for your help Thankyou Tahir Riaz Assistant Director (Systems) COMSATS Internet Services COMSATS Headquarters Building 9, Shahrah-e-Jamhuriat, G-5/2 Islamabad.

Hello List, i switched mailformat from mbox to maildir. Now i have a little problem with procmail/formail and headermanipulation of mails. Here an example: 0 * ^From.*gmx.de | (formail -t -I"procmail: gmx.de") >> /var/spool/mail/xxxxxx So, i can add some different headerlines for later evaluation and the result is attached to inbox. Now i have the problem that i don't know

Hello there, I have been trying to make the patch work for libwrap(TCP Wrappers) posted on http://dovecot.org/patches <http://dovecot.org/patches%20Patch%20of%201.1> Patch of 1.1 but could not get it work. Any help will be highly appreciated. After compiling and running it I get error "Error: login_tcp_wrappers can't be used because Dovecot wasn't built with

i need to catch the call hold event from my asterisk-java program. Im using net.sf.asterisk.*; for communicating with asterisk server. I need to get the call hold status on my java program . I can able to get the music on hold status but i cannot able to get the call hold status. The events like 1. HoldEvent , 2.HoldedcallEvent 3. UnHold event are not getting fired when the call hold is

Dear All I want to use asterisk for the following Senario and Need help to find a SAMPLE extension.conf Incoming call >>>>>>>>>>>Asterisk >>>>>>>>>>>>>>GSM Termination Gw first leg second leg What I want to do is putting first call leg on

Thanks Richard. This is exactly the answer I was looking for. I'm now assuming that Asterisk 11 was using it's equivalent "bridge_simple" but I was getting confused because the only bridge module I saw in modules.conf was bridge_softmix. When I upgraded to Asterisk13 that would have been the only bridge getting loaded at first. Is it expected that if bridge_softmix handled a

Hello, I have been trying to completely disable music on hold on my asterisk system. When a call is put on hold I do not want any music on hold, but I would like the remote user to get informed of this event (depending on the technology e.g. with a SIP reinvite and an SDP indicating the call is on hold). I have searched and tried out various approaches, but when putting the call on hold

On 2/23/19 4:19 PM, Joshua C. Colp wrote: > On Sat, Feb 23, 2019, at 11:04 AM, hw wrote: > > <snip> > >> >> directmedia is not explicitly enabled; I guess it's the default. >> >> Joshua basically says there is no way to control which ports are being >> used for SRTP because that it is "up the endpoint". Such endpoints, in >>

Hello List, i'm looking for a way to increase log-level for imap-processes. Pop3 Server writes all necessary infos about mails-transfered/deleted etc. Imap-Log writes just times and ip's of Userlogins. Is there a way to increase the imap-output? The Conf-Parameters i found did not really fit my requirements. Best result would be a collected log of transfering/deleting mails for one

Hi everybody We have a strange problem with several asterisk servers (Version 1.4.11) using PRI cards (tied to telco here in Belgium). Indeed we noticed that whenever a local user places an outgoing call through the PRI (and telco) to another IPBX (tied to telco using BRI or PRI), if the remote party places the call on hold, the caller hears the _local_ music on hold instead of the

On Mon, Jun 29, 2020 at 6:46 AM Joshua C. Colp <jcolp at sangoma.com> wrote: > On Sun, Jun 28, 2020 at 2:26 PM Dovid Bender <dovid at telecurve.com> wrote: > >> Hi, >> >> We have a box up and we are starting to see a lot of "Exceptionally long >> queue length queuing" in the logs. From all the research so far it seems >> like this leads to