search for: 183

...146 146 212 212 291 291 218 218 170 170 187 187 194 194 190 190 214 208 152 152 158 158 155 155 209 209 -9 -9 180 180 150 150 185 185 -9 -9 -9 -9 211 211 152 152 -9 -9 155 155 212 212 209 209 234 234 166 166 187 187 -9 -9 -9 -9 -9 -9 -9 -9 -9 -9 -9 -9 212 212 205 205 236 236 164 164 187 187 194 194 183 183 214 214 163 163 160 160 155 155 212 212 203 203 238 236 164 164 187 187 -9 -9 179 179 -9 -9 -9 -9 -9 -9 -9 -9 212 212 203 203 236 236 164 164 187 187 194 194 -9 -9 214 214 165 165 160 160 155 155 212 212 209 209 236 234 164 164 187 187 194 194 179 179 -9 -9 -9 -9 -9 -9 -9 -9 212 212 205 205 236...

...softphones sends a SIP message to the Asterisk. In this case the dialplan will execute the instruction ' exten => 2005,1,Dial(SIP/2000&SIP/2001&SIP/2002, 30) ' All softphones (2000, 2001 and 2002) will ring. These are proprietary softphones and all of then will reply with SIP 183 message. SIP 183 will contain SDP with media information. The question is: Will the caller receive SIP 183 from each callee? That is, will it receive 3 SIP 183 messages? It is important to the caller receives a SIP 183 message from each callee, because this caller needs to send early media (vide...

...>> >> Or should Asterisk create the ringback (Asterisk doesn't send any RTP >> package)? Or should the phone >> create the ringback itself because there is a 180 Ringing (even if it >> contains SDP)? >> >> I'm wondering: Why does Asterisk create a 183 to the extension >> containing SDP if the callee didn't >> provide any SDP? >> >> >> So many questions ... . Could somebody please shine some light on it? >> What's going wrong here? > > The core doesn't communicate whether progress includes...

...tyougood postfix/smtpd[17497]: name_mask: resource Sep 25 04:53:30 hostyougood postfix/smtpd[17497]: name_mask: software Sep 25 04:53:30 hostyougood postfix/smtpd[17497]: xsasl_dovecot_server_create: SASL service=smtp, realm=(null) Sep 25 04:53:30 hostyougood postfix/smtpd[17497]: connect from c-67-183-127-210.hsd1.wa.comcast.net[67.183.127.210] Sep 25 04:53:30 hostyougood postfix/smtpd[17497]: match_list_match: c-67-183-127-210.hsd1.wa.comcast.net: no match Sep 25 04:53:30 hostyougood postfix/smtpd[17497]: match_list_match: 67.183.127.210: no match Sep 25 04:53:30 hostyougood postfix/smtpd[17497...

...e caller. This behavior is completely correct, because there is no way to know which early media audio stream to pass back to the caller in a parallel call scenario (as in this case several endpoint may indicate session progress all at the same time). The question is why is asterisk still sending 183 session progress back to the caller if no audio is to be bridged before the 200 OK anyway? If 183 are not passed back to the caller, then at least a 180 Ringing that may come from another endpoint will cause the calling endpoint to generate local ringback. This won't happen if the caller has re...

Hi SamyGo. Thank you for the replay. So, let me explain it better: I knew that I could use something like " same = n,Dial(PJSIP/6001&PJSIP/6002) ". While every extension (called phones) rings and before anyone answers, SIP 183 messages will be sent to Asterisk from callees. If a called phone answer, the others will be hanged up. It is ok for me. I want to connect the caller just to the first called party that answers. Yes, it is some sort of ring group implementation where users are dialled and just the first one to answ...

[Resent due to non-descriptive subject line.] Hi folks When connecting two SIP users, is there any way to stop Asterisk from sending SIP 183 Session Progress messages, either globally or per-peer? Scenario as follows: Call from UA1 to Asterisk (UA2) to UA3. UA3 sends RTP before SIP OK to Asterisk (UA2). Asterisk (UA2) detects early audio from UA3 and sends 183 Session Progress with SDP to UA1. Instead I would like it to just sen...

Hi, I have a Asterisk that connects to the PSTN via a PRI. After Asterisk sends the setup message it immediately sends a 183 Session Progress. Is there a way I can change it so that it sends a 100 Trying instead? Because I am having some issues with a equipment where it does not play a busy tone as a result of sending a 183 Session Progress then the 486 Busy. Thanks Remi

...ific the latest issues can be seen by the snippet below on the Freefilesync log. Im trying to dial in the symlink operation, and any other remaining errors. This demonstrates two issues 1) Cannot write file Error Code 2: The system cannot find the file specified. (BackupWrite) 2) Error Code 183: Cannot create a file when that file already exists. (CreateSymbolicLinkW) The symbolic link setting I think I tried both ways, I think I have another toggle on the 3rd party app(ffs) which I need to also have in the correct setting, and to get the combination correct. Im not sure what the proble...

Dear all I have confusion how to asterisk genrate tone and what ringing code use default 180 or 183 i have setup asterisk with mediant 2000 with avaya [asterisk]-----[mediant 2000]--------[Avaya] when i call from avaya side to ---> asterisk i don't got ringback Sound so how to asterisk genrate tone for calling party is there any soution and what is the problem of this type of problem...

Hello everybody, I have 2 users connected on the same Asterisk server that are connected with 2 different Asterisk servers. For outgoing calls, one in receiving 183 Session Progress and the other not! Do you have any idea why? Thanks. I have tried to understand by myself and in their INVITE they have almost the same Allow and Supported SIP Headers The one that works: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replac...

...erface through which I connected an external PSTN line. I use it as carrier for VoIP calls. I can make successfully calls, but there's one problem, I receive 200 OK with SDP with delay (sometimes more than 30 seconds). So when I make a call through asterisk I receive intially: - 100 Trying - 183 Session Progress, with SDP when the called number respond, I start receiving RTP with voice, also the called receives voice from me, but only after a while asterisk sends 200 OK with SDP. I'm not sure if the problem is from asterisk or from the telephony provider (I think the provider). Is...

...Enviado: segunda-feira, 13 de julho de 2015 17:43 Para: Asterisk Users Mailing List - Non-Commercial Discussion Assunto: Re: [asterisk-users] RES: How to dial extensions asynchronous-sequentially ? All I can focus now is "the objective is to see if there is an way to deliver more than one SIP 183 message to the caller" 6001 has a song playing in 183 and 6002 has a "service unavailable" message, do you intend to deliver both of them simultaneously to the caller? I've seen multiple 183 Session Progress messages getting delivered to caller but what is your end game ? Play a...

Ok Mark Michelson. Thank you very much! You answer tells me that I was in the wrong path trying to access information from SIP 183 message. I need to find a way to let the callee pass information/data to the caller, even before accepting the call. That is, send data during the ringing time. And in my case, there will be more than one callee ringing at same time. As ASTERISK will not forward each SIP 183 message to the caller,...

...aling on our PSTN gateways to provide ringback to the PSTN network - this makes complete sense (gateway to PSTN gets a S: rg). The problem is that some of our older gateways don't support MGCP event signaling, so I was trying to get Asterisk to send all SIP calls to our switch with a "183" message so maybe our switch will cut the call through two-way immediately. Currently Asterisk is sending 180/ringing messages to our switch. Is there a way to have Asterisk send 183 messages to my peer defined in sip.conf ? Thanks, Dave

...y, but do not have this problem. We think it has something to do with >asterisk 1.6. The other asterisk systems are using 1.4. I have played >around with "progressinband" in sip.conf with now success. Whatever I set >progressinband to, it doesn't seem to change a thing. "183 Session >Progress" never seems to be called when looking at sip debug. It is only >when I use the Progress application before my dial command that I get the >"183 Session Progress" message in sip debug. > >We also have a Trixbox system using asterisk 1.6 that had the s...

Hello R help, I have this data frame M2[160000,5] with NAs, a simple example would be: set.seed(1234) M2<-expand.grid(ID=182:183, year=2012, month=1:3, day=1:3, KEEP.OUT.ATTRS=FALSE) M2 <- M2[with(M2, order(ID, year, month, day)),] #sort the data M2$value <- sample(c(NA, rnorm(100)), nrow(M2), prob=c(0.5, rep(0.5/100, 100)), replace=TRUE) M2: ID year month day value 1 182 2012 1 1 -0...

Hi. The ASTERISK wiki has a page showing the function PJSIP_HEADER(). However, it doesn't explain if such function works only over SIP INVITE messages or if it can be use, for example, to read headers from others types of SIP messages too. So, can I use PJSIP_HEADER to read the SIP 183 message header? Any hint will be very helpful! Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 (Brasil)

Hi all,

...a bit puzzled because i do not get include and hint to work with the new realtime enginge (cvs-head from 2004-12-09). other things (sipfriends and "normal" extensions) work perfect with the realtime engine. the entries in the static extensions.conf file i used before where: exten => 183,hint,SIP/snom220 exten => 183,1,Macro(stdexten,443,SIP/snom220,183) exten => 187,hint,SIP/zyx2000 exten => 187,1,Macro(stdexten,447,SIP/zyx2000,187) the entries in the table for realtime config look like this: SELECT context, exten AS ext, priority AS prio, app, appdata FROM extensions W...