search for: mohinterpret

...old -- this removes the need 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,...

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 peer B is taken......

...=> mysql,asterisk,iaxfriends voicemail => mysql,asterisk,voicemail --------- - Mysqldump from iaxfriends --------- INSERT INTO iaxfriends (name,type,phonenumber,username,mailbox,secret,dbsecret,context,regcontext,host,ipaddr,port,defaultip,sourceaddress,mask,regexten,regseconds,accountcode,mohinterpret,mohsuggest,inkeys,outkey,language,callerid,cid_number,sendani,fullname,trunk,auth,maxauthreq,requirecalltoken,encryption,transfer,jitterbuffer,forcejitterbuffer,disallow,allow,codecpriority,qualify,qualifysmoothing,qualifyfreqok,qualifyfreqnotok,timezone,adsi,amaflags,setvar) VALUES ('admin.my....

...using the very same MYSQL tables (and columns) with Asterisk 1.4.11 and things are working well. The questions I have are, since new configuration variables have been added into Asterisk 1.4, can I simply add columns in my MySQL sippeers table for things such as "videosupport" "mohinterpret", etc.? When a user upgrades, how would one know if all of the possible user/peer/friend variables listed in sip.conf can in fact be pulled from Realtime? I have consulted http://www.voip-info.org/wiki-Asterisk+RealTime+Sip but that table seems to be out of date. For example, "musico...

...Asterisk 1.4.14. Here'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 notifymimet...

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...

...=yes callwaiting=yes 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......

...;linear transmit gain (1.0 = no change) language=de ;set default language ;ulaw=yes ;set this, if you live in u-law world instead of a-law ;jb..... ;with Asterisk 1.4 you can configure jitterbuffer, ;see Asterisk documentation for all jb* setting available. ;mohinterpret=default ;Asterisk 1.4: default music on hold class when placed on hold. ; interface sections ... [ISDN1] ;this example interface gets name 'ISDN1' and may be any ;name not starting with 'g' or 'contr'. ;Use one interface sec...

...dd this code in extensions.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...

...pport: no maxcallbitrate: 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 dtlsc...

...threject = no autodomain = no callevents = no canreinvite = yes checkmwi = 10 compactheaders = no defaultexpiry = 120 dumphistory = no externip = 216.26.109.22 g726nonstandard = no jbenable = yes jbforce = no jblog = no localnet = internal subnet maxcallbitrate = 384 maxexpiry = 3600 minexpiry = 60 mohinterpret = default nat = yes notifyringing = yes pedantic = no progressinband = never promiscredir = no realm = asterisk recordhistory = no registerattempts = 0 registertimeout = 20 relaxdtmf = no sendrpid = no sipdebug = no t1min = 100 t38pt_udptl = no tos_audio = none tos_sip = none tos_video = none trust...

...ideosupport: no maxcallbitrate: 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 dtlsc...

...;linear transmit gain (1.0 = no change) language=de ;set default language ;ulaw=yes ;set this, if you live in u-law world instead of a-law ;jb..... ;with Asterisk 1.4 you can configure jitterbuffer, ;see Asterisk documentation for all jb* setting available. ;mohinterpret=default ;Asterisk 1.4: default music on hold class when placed on hold. ; interface sections ... [ISDN1] ;this example interface gets name 'ISDN1' and may be any ;name not starting with 'g' or 'contr'. ;Use one interface sec...

...fig) authname=089xxxxx secret=xxxxxxxx (as stanaphone give in the sip config host=sip.stanaphone.com allow=all (tried that since the softphoen uses pcm when it works - no change) allow=g729 allow=gsm dtmfmode=rfc2833 insecure=very canreinvite=no qualify=yes nat=yes port=5060 context=richardincoming mohinterpret=better I don't believe that the extensions.conf is a problem since I have other voips going to the same 8800 extension and being handled right. What I get in the console on an incoming call to the stanaphone number is. -- Executing [8800 at richardincoming:1] NoOp("SIP/089xxxxx-0...

...ip=cs3 ; 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...

...constantssrc: NULL contactpermit: 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...

...number from the range. 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 ;...

Hello list, what musiconhold class has priority : - field "musiconhold" of the SIPaccount and field "musiconhold" of a queue or - Set(CHANNEL(musicclass)=) ?? Kind regards, Jonas. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110201/b27a0534/attachment.htm>

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: >> >>