similar to: sip_reg_timeout problem

Hello, I have trouble to enable a sip trunk with a CIRPAK. CIRPAK support answer that's there parameter are unvalid : a=silenceSupp:off - - - - is not standard and not working with cirpak - to be remove m=video 13072 RTP/AVP no video, how to remove it ? my extension.conf : exten => _6X.,1,Dial,SIP/${EXTEN:1}@x.x.x.x Regards, -- Arnaud Pignard (apignard@frontier.fr) Frontier Online -

For make outgoing call, i setup 0. However 0 is write in the cdr dst field. Is there a way to remove it when asterisk send it to cdr_mysql ? exten => _0X.,1,Dial,SIP/${EXTEN:1}@mygateway I just want have in cdr dst = ${EXTEN:1} This don't work : exten => _0X.,1,SetVar(EXTEN=${EXTEN:1}) exten => _0X.,2,Dial,SIP/${EXTEN}@mygateway Use another variable still record ${EXTEN} --

For make outgoing call, i setup 0. However 0 is write in the cdr dst field. Is there a way to remove it when asterisk send it to cdr_mysql ? exten => _0X.,1,Dial,SIP/${EXTEN:1}@mygateway I just want have in cdr dst = ${EXTEN:1} This don't work : exten => _0X.,1,SetVar(EXTEN=${EXTEN:1}) exten => _0X.,2,Dial,SIP/${EXTEN}@mygateway Use another variable still record ${EXTEN} --

Hi ! I have a TDM40B and i try to use it connected to modem for incoming call data transfert. I have no problem to use it with a phone and a talk communication work fine. But when we try to use with modem, with most modem, we got data carrier for few seconds and channel hungup. < [ TYPE: Null Frame (4) SUBCLASS: N/A (3) ] [Zap/4-1] -- Zap/4-1 is ringing << [ TYPE: Null Frame

Hi, Since around a week i have one asterisk server how stop responding randomely. CVS HEAD with RealTime engine used. The debug log only write "Failed to grab lock, trying again..." until i stop Asterisk. No more activity for IAX or SIP channels (no log...). CLI still responding. When i try to stop asterisk (stop now or crtl+C) nothing happen and cli die but asterisk still running.

Hi Just received my spanky new TE405P today to replace my Cisco gateway... After much fiddling (I forgot to switch it to E1) I got it to work and everything "seems" to work perfectly on our ISDN PRI. If I dial-in from the PSTN to a SIP phone, the call goes through and if I hangup either the SIP phone or the remote end, the call gets disconnected and destroyed However, if I dial-in

Hi, I have a partial working installation with zaphfc. Incoming call : For incoming call, seems work fine. But the sound is very bad with bounce short crashing sound. Same sound with echo cancel off or on. SDA work fine. Another problem, it's seems that's zaphfc don't reset correctly the line. I have one of my D channel how was busy even after stop communication. Outgoing call :

I am receiving this message on my asterisk server and I have commented out 5748150837 in my sip.conf file but it keeps showing this message on the server. Dec 10 07:59:31 NOTICE[30448]: chan_sip.c:5267 sip_reg_timeout: -- Registration for '5748150837@69.25.143.141' timed out, trying again (Attempt #1546) any ideas? -------------- next part -------------- An HTML attachment was

Hello, I have a problem with an LAN-Server behind an NAT-router. Asterisk Version 1.2.1 or 1.2.2 doesnt matter 10 minutes after starting Asterisk I loose all registrations at external SIP-proxys. The reason seemed to be that Asterisk send every second an request to every sip-proxy "Request: OPTIONS sip:sip.domain.tld". Every request is responded by the sip-proxy. After some minutes

Hello, I can?t compile "iaxclient", because one needs to compile the new version "wiax.dll". I tried to compile it under MinGW/Cygwin, but I had the messages like: cc -I. -Igsm/inc -Iportaudio/pa_common -Iportaudio/pablio -Iportmixer/px_common -Ilibspeex/include -g -O2 -DSPEEX_PREPROCESS=1 -DNEWJB -Ilibiax2/src -IAXC_IAX -DLIBIAX -DSPEEX_EC=1 -DWIN32 -DBUILDING_DLL -c

Hello, Is there a reason for the following behaviour? > mean(c("1","2","3")) [1] NA Warning message: In mean.default(c("1", "2", "3")) : l'argument n'est ni num?rique, ni logique : renvoi de NA But: > var(c("1","2","3")) [1] 1 And also: >

Hello, I?ve been using Asterisk 1.2.18 for a while, and today, with no apparent changes, I started receiving these messages: Aug 31 13:26:57 NOTICE[27528]: chan_sip.c:5495 sip_reg_timeout: -- Registration for 'user at sipserver' timed out, trying again (Attempt #19) All trunks and extensions went to: sipserver:5060 user 120 Request Sent 011

I have a time series of 1095 data corresponding to a daily data of three years. I want to know how to use ma(timeserie, order=??, centre=??) to detect the trend: which order is suitable and what is the difference between centre= true or false. How to avoid these errors: 1-Error in timeserie - trend : ? argument non num?rique pour un op?rateur binaire="non-numeric argument for a binary

Thks for your answer, here is an exemple of what i do with the errors in french... > tmp [1] 200 150 245 125 134 345 320 450 678 > beta18 Erreur : Objet "beta18" not found //NORMAL just to show it > eta [1] 500 > func1<-function(beta18) dweibull(tmp[1],beta18,eta) > func1<-func1(beta18) * function(beta18) dweibull(tmp[2],beta18,eta) Erreur dans dweibull(tmp[1],

Hi everybody, I've got a very strange problem: As far as I remember I didn't change anything on my Asterisk side. I have 2 SIP providers to which I can place outbound calls. Today I noticed that outbound calls to provider "inode" fail (and inbound from this provider too). On the CLI I get every 20 seconds following messages: Nov 9 20:01:07 NOTICE[952]: chan_sip.c:5422

I can not register via sip to iptel or sipgate and do not see sip into ethereal. I do not unterstand why thats Wudu .. but i am new to asterisk and sip. I am behind a susefirewall2 but asterisk even do not register if it is down. The asterisk is running onto the machine witch is connected to the internet. No answer seems coming back from iptel (sip debug in asterisk). Ports are open (5060,

I think median() behaves as designed: As long as the argument can be ordered, the "middle observation" makes sense, except when the middle falls between two categories, and you can't define and average of the two candidates for a median. The "sick man" would seem to be var(). Notice that it is also inconsistent with cov(): >

Thanks for taking a look greg and hank. This seems to be getting bettre everyday..help please My sjphone is running on the same box as asterisk...i believe then the red hat firewall should not be a problem. Whenever i dial from CLI i get ######### Executing Goto("OSS/dsp", "default|s|1") in new stack -- Goto (default,s,1) -- Executing Wait("OSS/dsp",

Hi All, I have just installed Asterisk and am trying to connect it to a SIP account that I currently have with www.voiptalk.org but without any success. Although I know that voiptalk do provide asterisk accounts I don't want to convert the SIP account until am happy that it's gonna work for me. The asterisk box is currently behind a firewall and the following ports are being forwarded

Hi All, I am trying to setup another sip trunk in addition to what I am already using. The sip provider we are using right now gives you your username as your email address. So IE. If my email is james@james.com.... that is my username . Now... When I put this in the sip.conf file I have found that Asterisk is not able to parse it correctly and instantly goes to the email server to authenticate