similar to: asterisk 11.7.0: Delayed audio

Hello! Just installed asterisk 13.2.0 and see many such messages in log, I see them in console during calls, really something like this: -- Executing [6166 at kanbaikal:2] Dial("OOH323/kanbaikal-6", "SIP/6166 at asterisk") in new stack == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Called SIP/6166 at asterisk > 0x7fa9d4007660 --

i noticed that when i active the voicemail in the IP-phone where the number 0033149xxxxxx is configured i can call this number without issue Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Called SIP/FD/0033149xxxxxx == Begin MixMonitor Recording SIP/101-0000010d -- SIP/FD-0000010e is making progress passing it to SIP/101-0000010d > 0x2b393cfc2610 -- Probation passed

Hi there, I'm facing the decision if it would be possible to transform several more or less complex pdf files into an R Table-Format or if it has to be done manually. I think it would be a impudent to expect a complete solution, but I would be grateful if anyone could give me an advice on how the structure of such a R-program could look like, and if it's possible in general. Here

Hi, with canreinvite=no and directmedia=no I and getting the message in the logs for all calls "switching from simple_bridge technology to native_rtp" -- Executing [102 at mkg:1] Dial("SIP/101-00000017", "SIP/102") in new stack == Using SIP RTP CoS mark 5 -- Called SIP/102 -- SIP/102-00000018 is ringing -- SIP/102-00000018 answered SIP/101-00000017

05.03.2015 11:42, Dmitry Melekhov ?????: > 05.03.2015 11:29, Dmitry Melekhov ?????: >> Hello! >> >> Just installed asterisk 13.2.0 and see many such messages in log, I >> see them in console during calls, really something like this: >> >> >> -- Executing [6166 at kanbaikal:2] Dial("OOH323/kanbaikal-6", >> "SIP/6166 at

NAT endpoint calling local endpount - switching to native_rtp then no audio, both of them have direct_media=no, Verbose log: -- Executing [99 at dialmap:1] AGI("PJSIP/304-00000022", "/pbx/agi.php") in new stack -- Launched AGI Script /pbx/agi.php -- AGI Script Executing Application: (Dial) Options: (PJSIP/99/sip:99 at 192.168.1.73:5060,20) -- Called

On 2017-04-18 02:42 AM, Pete Mundy wrote: > Try this: > > asterisk -r > core set verbose 10 > [get user to trigger fault] > [examine console output, and post to list if still unclear] > > If you don't solve it yourself, then we'll be able to help further once > we've seen the output. I can't see much more than at my previous debug level but here it is

tnaks for your response but the number dialed exist and i can call this number when i configure the trunk directly in x-lite and i call call also this number from my cell phone . any help thanks and regards 2015-03-25 12:59 GMT+00:00 Matthew Jordan <mjordan at digium.com>: > On Wed, Mar 25, 2015 at 7:35 AM, Salaheddine Elharit > <salah.elharit200 at gmail.com> wrote: > >

Hi, I use a simple scheme: SIP video phone A (h264/Asterisk 1.8.11) <---IAX2 trunk----> SIP video phone B (h264/Asterisk 11.7.0) When calls from A to B and vice versa drop on pickup. On B side: [Oct 24 16:33:49] DEBUG[15590][C-00000012] res_rtp_asterisk.c: Setting the marker bit due to a source update [Oct 24 16:33:49] DEBUG[15590][C-00000012] res_rtp_asterisk.c: Setting the marker bit

hello list, i have asterisk 11.15.0 and i have some trunks sip from my provider we have some ip phone astra 6731i each Ip-phone is configured with trunk and we call no ihave configured another trunk from the same provider in my asterisk i can call all numbers just the numbers are configured in thses ip phones. but when i configured the same trunk in x-lite i can call theses ip-phones without

Hi list, I'm working with gamm models of this sort, using Simon Wood's mgcv library: gm<- gamm(Z~te(x,y),data=DATA,random=list(Group=~1)) gm1<-gamm(Z~te(x,y,by=Factor)+Factor,data=DATA,random=list(Group=~1)) with a dataset of about 70000 rows and 110 levels for Group in order to test whether tensor product smooths vary across factor levels. I was wondering if comparing those two

The Asterisk Development Team is pleased to announce the release of Asterisk 10.1.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 10.1.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in

Hi, I am working with gam models in the mgcv library. My response variable (Y) is binary (0/1), and my dataset contains repeated measures over 110 individuals (same number of 0/1 within a given individual: e.g. 345-zero and 345-one for individual A, 226-zero and 226-one for individual B, etc.). The variable Factor is separating the individuals in three groups according to mass (group 0,1,2),

Dear list, I have another (again possibly boneheaded) puzzle about importing, again encapsulated in a nearly trivial package. (The package is posted at <http://www.math.mcmaster.ca/bolker/misc/coefsumtest_0.001.tar.gz>.) The package consists (only) of the following S3 method definitions: coeftab <- function(object, ...) UseMethod("coeftab",object) coeftab.default <-

Dear list, I am fitting a logistic multi-level regression model and need to test the difference between the ordinary logistic regression from a glm() fit and the mixed effects fit from glmer(), basically I want to do a likelihood ratio test between the two fits. The data are like this: My outcome is a (1,0) for health status, I have several (1,0) dummy variables RURAL, SMOKE, DRINK, EMPLOYED,

Hi, I configured Buddy Watch function on my Polycom IP 601. It works well, until I make a reload of Asterisk. After reload, if I give the "show hints" command in Asterisk's CLI, it says that there are no watcher for the extensions that I configured. Before the reload in the CLI appears: -= Registered Asterisk Dial Plan Hints =- 3002 : SIP/3002 State:

Dear all, I found a problem with 'lme4'. Basically, once you load the package 'aod' (Analysis of Overdispersed Data), the functions 'lmer' and 'glmer' don't work anymore: library(lme4) (fm1 <- lmer(Reaction ~ Days + (Days|Subject), sleepstudy)) (gm1 <- glmer(cbind(incidence, size - incidence) ~ period + (1 | herd), family = binomial, data

I have used every asterisk 11.8.X version. Have not had an issue till 11.8.0 and 11.8.1 When I use ConfBridge I am attempting to put all participants in MUTE mode and just one talker... However, since 11.8.0 I am hearing feedback in the announcement like the channel is not really muted. I dropped back to 11.7.0 and I hear no feedback. Has something changed - or - am I not correctly setting up

Hi all, I have problem for receiving fax from multiple page fax that sent from fax machine (analog). The error is : WARNING T.30 Page did not end cleanly This is my dialplan [inboundfax] exten => s,1,NoOp(**** FAX RECEIVED from ${CALLERID(num)} ${STRFTIME(${EPOCH},,%c)} ****) exten => s,n,Set(FAXOPT(ecm)=yes) exten =>

Dear all, I am running glmer() in R. How can I retrieve the estimates of fixed effects and the variance of the random effects from the result? Thank you so much. Joe ___________________________________________________ ±zªº¥Í¬¡§Y®É³q ¡Ð ·¾³q¡B®T¼Ö¡B¥Í¬¡¡B¤u§@¤@¦¸·d©w¡I [[alternative HTML version deleted]]