similar to: Ring + Music on Hold in the same call

Hi all, we have a problem with a reinvite sent by our SIP provider to change audio codec due to the recognition of a fax tone. After that the SIP call session has been established (INVITE and 200 OK) we have the following codec situation: UAC ASTERISK UAS | ASTERISK UAC PROVIDER g711 <----------------------> g711

On Sat, Jun 18, 2011 at 6:40 AM, Larry Moore <lmoore at starwon.com.au> wrote: > On 18/06/2011 5:36 AM, Matteo Campana wrote: > >> >> Inviato da iPhone >> >> Il giorno 16/giu/2011, alle ore 16:37, Eric Wieling<EWieling at nyigc.com> >> ha scritto: >> >> We experience the same thing. The solution we use is to not change >>>

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Hello list, using asterisk 1.4.30. When setting up the MySQL table 'musiconhold' as described in http://www.voip-info.org/wiki/view/Asterisk+config+musiconhold.conf , what is the meaning of the fields : `*digit*` char(1) NOT NULL default '', `*sort*` varchar(16) NOT NULL default '', and what are there default values ?! What is the default value of :

Dear all I wrote a routine. At the end of each cycle of the loop I would like to save the result (plot) in a postcriptfile. Of course if I just use dev.print in the following way: dev.print(device=postcript, 'c:/Rfigures/plot_1.ps") I overwrite my results with the second cycle of the loop. I suppose there is a way to define the file name so that several plots are

Using control_work instead of config_work as the 3rd argument to container_of results in an invalid portdev pointer. Indeed, the work structure is initialized as below: INIT_WORK(&portdev->config_work, &config_work_handler); It leads to a crash when portdev->vdev is dereferenced later. This bug is triggered when the guest uses a virtio-console without multiport feature and

Using control_work instead of config_work as the 3rd argument to container_of results in an invalid portdev pointer. Indeed, the work structure is initialized as below: INIT_WORK(&portdev->config_work, &config_work_handler); It leads to a crash when portdev->vdev is dereferenced later. This bug is triggered when the guest uses a virtio-console without multiport feature and

Asterisk 1.4.6 at FreeBSD6.2-RELEASE Client hears pure silence when waiting for call answer. Music on hold stops when transferer pics a number and client doesn't even hear ringing. Is this normal behaviour? How to change this? Log says everything, MOH should stop after call pickup, not before Dial. -- Executing [113 at firma:1] Dial("SIP/zytek-08737000",

Hi, I'm having random hangup problems with zap channels. If I turn busydetect off in Zapata.conf, * fails completely to detect a user hangup in the middle of a script. On the other hand, if I turn it on, everything works much better, but long calls tend to be hung up without a motive. Any other parameter that I can try? Any #define that I can tweak and recompile? Will callprogress

Hi, All Is there a provision for "AbsoluteTimeout" application to notify called and calling party of the reason why the call suddenly ended? This way, the parties will be much better informed, hence they will/should not think that their VOIP/telco provider(s) are providing bad service. Ta SJ

Sorry for the fragmented messages from me - one last thing I forgot to ask in my last post. When incoming calls come to us, our PSTN line is picked up almost immediately - and then asterisk will proceed to dial the SIP extensions. During this time the caller hears dead slience - obviously not very good as some would think the line just went dead and hang up. I have toyed with the idea of playing

Hi, I have trouble getting MOH to work after an upgrade from asterisk 1.4 to 1.6.1.4. The call goes on hold, MOH is started, and then stops right away. Here are the files both of type .raw: Tsunami*CLI> moh show files Class: default File: /etc/asterisk/musiconhold/Fr?d?ric Chopin - Polonaises Op. 40-2 File: /etc/asterisk/musiconhold/Fr?d?ric Chopin - Polonaises Op. 40-1 These files

Matteo Brancaleoni wrote: >Hi. > >I'm experiencing a strange issue with *. >I have a dev kit, aka a T100P + a zhone cb. > >Sometimes, on certains phones (on the fxo ports >of the cb) , when the phone rings, * detect >it as answered after the first ring, even >if no one is at the phone! > >The result is that on the other party (which >called the phone) hears

Dear list, I have been using asterisk for some time now. However I have never used it with any of the digium or compatable cards (Purely used for SIP). I understand that for using Meetme, I need to have a timing device, which could either be hardware or zrdummy etc (I am not using any right now). Can someone tell me if the timing device is needed for voicemail and other applications too?. I am

I am trying to get the handytone 286 to make a very simple call to * and having problems. It registers with * just fine, but when I place a call (to echo test, for example), the RTP stream seems to have problems opening. Here is there error I get in *: WARNING[98311]: File chan_sip.c, Line 464 (retrans_pkt): Maximum retries exceeded on call 20d1c411-e210-5f3d-3f88-19035c8fcb26@192.168.2.6 for

On late august, there was a thread about setting up some meetme conferences to be able to follow Astricon remotely. This indeed could be nice for those that can't attend for various reason. And of course is a demonstration of Asterisk capabilities... :) (Astricon without a remote conference for guest is like a big it expo without internet connections...) I have some bandwidth here, so can

Hello, We are working on a new Asterisk installation and have run into some problems related to playing MusicOnHold for a caller when they have been placed on hold by an agent, that took the call from a queue. A. When pressing the HOLD button on SNOM 190 and Grandstream BudgeTone SIP phones, MusicOnHold works fine when making inbound or outbound direct calls by extension. Music starts to play

~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ CLI Output : ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ vicidialnow*CLI> == Parsing '/etc/asterisk/manager.conf': Found == Manager 'sendcron' logged on from 127.0.0.1 == Parsing '/etc/asterisk/manager.conf': Found == Manager 'sendcron' logged on from 127.0.0.1 == Manager 'sendcron' logged off from

Hi all, I'm trying to share a folder between a host and a guest using VirtFS. The source is NAS mounted on the host and QEMU runs as unprivileged user. The only way for me to use the folder inside the guest is to set the access mode to 'squash' but all file operations are performed by the user running QEMU. If I ran QEMU as root, the passthrough mode works as expected. Is there

I was wondering what are the values for sending dmtf via sip info. I mean, when I use dtmf relay via sip info, the sip/sdp message contains a Signal=X where X is the dmtf. That's ok for dtmf 0-9 . but what when dtmf is * or # ? we must send signal=# ? I ask that because I noticed that budgetones phone sends out * as signal=10 and # as signal=11 . but asterisk don't detect them, 'cause