search for: mohsuggest

Hello list, what is the difference between mohsuggest & mohinterpret when defining a SIP peer ?! If a certain SIP peer puts another channel on hold, what field then determines the moh class that Asterisk will choose to play to that channel ? If I take the test and call from peer A to peer B, and peer A puts peer B in hold, then the class of...

I'm wondering if moh is still broken in Asterisk 1.4 beta3. In Asterisk 1.2, when a callee put a caller on hold, the musiconhold class that was played was not the one the callee wanted the caller to hear, but something else. Even after using mohsuggest in Asterisk 1.4, it still appears that this is not working correctly. Here's the results of a simple test: CASE CALLER CALLEE HOLDER HOLDER HEARS MOH ------------------------------------------------------ 1 3254101 3254102 3254101 moh1 2 3254101 3254102 325410...

...the database, or the new account 444 ??? Below are the conf files and verbose output. Thank you very much for your help :) --------- - iax.conf --------- [general] bindport=4569 delayreject=yes language=fr autokill = yes calltokenoptional = 0.0.0.0/0.0.0.0 minregexpire = 60 maxregexpire = 500 mohsuggest=default careinvite=no rtcachefriends=yes [444] type=friend host=dynamic context=special secret=iop --------- - extconfig.conf: --------- [general] [settings] iaxusers => mysql,asterisk,iaxfriends iaxpeers => mysql,asterisk,iaxfriends voicemail => mysql,asterisk,voicemail --------- -...

...d of streaming audio from server A to server B while B1 is on hold, which in my scenario is a good thing. I post to the list trying to get peer feedback to my initial tests. The configurations I mention are always applied to both servers A and B. 1. If I set mohinterpret=passthrough + mohsuggest=default in the [general] section of iax.conf the "local music on hold" never works. Results: bad - A1 calls B1, B1 puts A1 on hold, A1 gets B's music bad - A1 calls B1, A1 puts B1 on hold, B1 gets A's music bad - B1 calls A1, A1 puts B1 on hold, B1 g...

...l realms ? For now, I must add the realm prefix in the sip username of chan_sip. For example: [lg_2540] amaflags = default call-limit = 10 host = dynamic language = en_US context = lg_default callerid = "LG" <2540> secret = XXXXXXXXXXXXXXXXXXXXXXXXXX type = friend subscribemwi = no mohsuggest = default qualify = yes fromdomain=lg.allocloud.com fromuser=2540 If I use only [2540] as section name, I'll have a clash on the same Asterisk. Thanks for your answers. -- Ludovic Gasc (GMLudo) http://www.gmludo.eu/ -------------- next part -------------- An HTML attachment was scrubbed... UR...

I am having a strange problem with MOH. Say I have two users, A and B. I can set MOH in the extension for B and if A calls B and B hits hold, A will hear B's hold music. If however A hits hold, it goes to the default music. If I pull the setmusiconhold from extensions.conf and use musicclass in sip.conf under the peer A, I get the same thing. Peer A has musicclass set and A calls B and B

Hello! I use asterisk with TE420 as PRI switch for two channels : ;panasonic uplink group=3 context=panasuplink ; relaxdtmf=yes ; immediate=yes rxgain=0.0 txgain=0.0 mohsuggest=default jbenable = no ; jbenable = yes ; jbmaxsize = 200 ; display_send=name_initial display_send=name display_receive=name ; display_receive= channel=>63-77,79-93 ;panasonic group=4 priindication = outofband...

...re's a section from my sip.conf for my test phone: [general] context=internal allowguest=no allowoverlap=no allowtransfer=yes notifyhold=yes bindport=5060 bindaddr=0.0.0.0 srvlookup=yes pedantic=yes vmexten=9998 at internal ;vmexten=*97 disallow=all allow=ulaw allow=ilbc mohinterpret=default mohsuggest=default language=en useragent=TCTC PBX ;dtmfmode = info fromdomain=10.10.60.253 ;relaxdtmf=yes [15] username=15 host=dynamic type=friend context=internal secret=edited-out subscribecontext=internal dtmfmode=rfc2833 ;defaultip=10.10.60.246 mailbox=15 ;subscribemwi=yes notifymimetype=text/plain che...

Hello list, I have defined a new MoH-class in musiconhold.conf : [default] mode=files directory=/var/lib/asterisk/moh random=yes ; *[106002] mode=files directory=/var/lib/asterisk/moh/106002 random=yes* In sip.conf I have this commented out : ;mohinterpret=default ;mohsuggest=default Asterisk sees these moh-classes and files : vps2301*CLI> moh show classes Class: default Mode: files Directory: /var/lib/asterisk/moh Class: 106002 Mode: files Directory: /var/lib/asterisk/moh/106002 vps2301*CLI> moh show files Class: default File: /var/lib/...

...usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes immediate=no echocancel=yes echocancelwhenbridged=yes echotraining=yes callgroup=1 pickupgroup=1 mohinterpret=default mohsuggest=default overlapdial=yes group=1 signalling = pri_cpe channel => 1-15,17-31 context = default | I would be gratefully, if you have an idea or some advices to me. Thanks ! -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.d...

...ons.conf my chan_Skype.conf [Account] secret=XXXXXX context=from-pstn exten= Account disallow=all allow=g729 allow=alaw allow=slin allow=ulaw auth_policy=accept buddy_presence=yes direction=both ;auth_policy=ignore buddy_autoadd=true ;buddy_presence=no mohinterpret=default ;mohsuggest=none Regards Khaled Chehab NGN Eng. Description: xplorium Operations Office - Lebanon Office : +961 1 868686 ext 115 Mobile: +961 3 045212 E-mail: <mailto:kchehab at xplorium.com> kchehab at xplorium.com MSN ID :KhalidChehab at hotm...

...e: NULL mailbox: NULL regexten: NULL fromdomain: testers.com fromuser: NULL qualify: NULL defaultip: NULL outboundproxy: PU.BL.IC.IP contactpermit: NULL contactdeny: NULL fullname: NULL cid_number: NULL callingpres: NULL mohinterpret: NULL mohsuggest: NULL hasvoicemail: NULL subscribemwi: NULL vmexten: NULL rtpkeepalive: NULL directrtpsetup: yes dtlsenable: yes dtlsverify: no dtlsprivatekey: /etc/asterisk/keys/asterisk.pem dtlssetup: actpass dtlscertfile: /etc/asterisk/keys/asterisk.pem dtlscafile: /etc/asterisk/...

...bitrate: NULL mailbox: NULL regexten: NULL fromdomain: testers.com fromuser: 660 qualify: NULL defaultip: NULL outboundproxy: 1.1.1.1 contactpermit: NULL contactdeny: NULL fullname: NULL cid_number: NULL callingpres: NULL mohinterpret: NULL mohsuggest: NULL hasvoicemail: NULL subscribemwi: NULL vmexten: NULL rtpkeepalive: NULL directrtpsetup: yes dtlsenable: yes dtlsverify: no dtlsprivatekey: /etc/asterisk/keys/asterisk.pem dtlssetup: actpass dtlscertfile: /etc/asterisk/keys/asterisk.pem dtlscafile: /etc/asterisk/...

...; Sets TOS for SIP packets. tos_audio=ef ; Sets TOS for RTP audio packets. canreinvite=no dtmfmode = rfc2833 notifyringing=yes limitonpeers=yes callcounter=yes [basic-phone](!) type=friend context=from_internal_phones nat=no qualify=yes host=dynamic mohinterpret=default mohsuggest=default call-limit=20 callgroup=1 pickupgroup=1 [21](basic-phone) secret=mypassword [22](basic-phone) secret=mypassword [200](basic-phone) secret=mypassword And here is a trace of a call coming in through the IAX trunk, ringing internal sip phones 21 and 22, while I try to pick it up from 200...

...tactpermit: NULL contactdeny: NULL usereqphone: NULL textsupport: NULL faxdetect: NULL buggymwi: NULL auth: NULL fullname: NULL trunkname: NULL cid_number: NULL callingpres: NULL mohinterpret: NULL mohsuggest: NULL parkinglot: NULL hasvoicemail: NULL subscribemwi: NULL vmexten: NULL autoframing: NULL rtpkeepalive: NULL call-limit: NULL g726nonstandard: NULL ignoresdpversion: NULL allowtransfer: NULL dynamic: NULL path...

...e. the chan_dahdi.conf [channels] usecallerid=yes callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes callgroup=1 pickupgroup=1 useincomingcalleridondahditransfer = yes mohinterpret=default mohsuggest=default #include dahdi-channels.conf switchtype = euroisdn signalling = bri_cpe group = 0 channel => 1,2,4,5,7,8,10,11 signalling=fxo_ks group = 0 channel =>13-28 the dahdi-channels.conf just on of the channels which I need a public phone number instead of 39XXX50 ;;; line="23 OP...

Hello again! I just got a SIP account and it seems - from a config on the net -, that I've configured it correctly. But I get no call to the outside. Registration was OK. I tried: channel originate sip/1/echo at iptel.org Application ... I see the channel active for a while, but no call gets established. In my config I have defined the section [iptel] for the outgoing call and I

PRI debug of the entire call would be great, also, switchtype would be awesome as well. Thanks! Matthew Fredrickson On Thu, Mar 24, 2016 at 4:07 PM, Carlos Rojas <crt.rojas at gmail.com> wrote: > Hi > > Did you activate the pri debug on the cli asterisk? > > On Thu, Mar 24, 2016 at 12:59 PM, Carlos Chavez <cursor at telecomabmex.com> > wrote: >> >>

...ix = 0711 unknownprefix = signalling=bri_net_ptmp usecallerid=yes callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=yes echotraining=100 mohinterpret=default mohsuggest=default callerid = asreceived immediate=no overlapdial=yes facilityenable=yes callprogress=yes group=1 context=isdn1 channel => 1-2 EOF /etc/asterisk/extensions.conf: [default] exten => _X.,1,NoOp(${EXTEN}) [isdn1] exten => _X.,1,Dial(SIP/${EXTEN}@sipgate,30,trg) exten => _X.,n,Han...

Hi All, Sorry if this has been covered already, but I don't tend to follow this list as close as I should these days. Problem is that if a call comes in to a queue without option 'r' specified - moh plays as expected. Now, when that call is answered, all is fine. Trouble comes when that person then puts the caller on-hold. No moh is heard by the caller (in fact, they get silence).