similar to: double ringing tone on asterisk 1.2 (workaround)

While I wait for the call to be answered I hear a "double ringing tone", like : expected tone : tuuu tuuu tuuu tuuu what I hear : tuuu tuuu tuuu tuuu tuuu tuuu tuuu tuuu the second "tuuu" I think is generated somewhere and not "true", since it sounds slightly different and the lambda between the first and the second is always

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 This is slowly driving me nuts! I have several Cisco 7960s with SIP 8.2/7.5 fw connecting to Asterisk 1.2.5 driving a TE110P on a BT EuroISDN PRI line. On all outgoing calls I get a double ring tone (UK style + US style). I also have a DECT phone on a Sipura SPA-3000 configured with UK tones. This gives me a double ring of UK + UK, so this

I've an asterisk 1.2 connecting to a quescom gateway via SIP, the caller (asterisk) can hear the called, but the called hears nothing. Since both machines are on public ip, what other problem can it be ?

Is there a way to pass variables/arguments to the h extension ? for example : [default] exten => _1098933X.,1,NoOp(CARRIER TWT->TIM, EXTEN: ${EXTEN}}, SIPCALLID: ${SIPCALLID}, SIPDOMAIN: ${SIPDOMAIN}) exten => _1098933X.,2,SetVar(_PROVA="bla") [lot of stuff, agi, goto, tricks and magic that happens] exten => _1098933X.,10,Dial(${CHAN_DEST},,L(3600000:3599900)) <-

let's suppose I have this dialplan : exten => _X.,1,Playtones(ring) exten => _X.,2,Dial(CAPI/contr1/${EXTEN},,g) exten => _X.,3,AGI(update) where "update" updates some db tables we have based on the type of extension Now, from the wiki : If the /g/ option is specified, and the called party hangs up before the calling party, then Dial exits with a return code of 0 to

I have strange peaks of machine load on my asterisk servers, looking at top the load is very high even if cpu usage is low and no swap memory is used. This happens on all the machines, some of them have asterisk, mysql, agi and digium cards on them, so I thought I was only asking too much, but yesterday I noticed the same behaviour on an asterisk machine with only two digium in it, no other

I've the following installation : |asterisk client| --- > |asterisk server| --- > |other asterisk server| all the connections are made in IAX, the client and first server allows 711 and 729 the other server only allows 729 since it has low bandwidth at disposal all the numbers but a few are routed to a digium card in the first server, the others are routed to the other server, this

Hi, I?m trying to get app_festival to work. I got the source from the Debian woody package of festival-1.4.2 and applied the patch that came with * sources it applied fine; then I made the debian package and installed it. I have this on extensions.conf: exten => 6700,1,Festival(Hi there how are you doing ?) When I dial 6700 I hear nothing and then * hangups: -- Executing

I'm confused a bit about the following and was hoping to get some answers on this group - What is exactly implied when we say asterisk can connect to a PSTN. Does it mean connecting to the PSTN via PRI/T1/E1? If yes, then I assume asterisk does not need to do any SS7 signaling and all it does (playing the role of a PBX) is to connect to a Class 5 Switch at the CO. Is this a correct statement?

Some of you might already know that we are releasing a new phone, snom 360. To make the phone well-known and stable, we have made a soft phone version out of it and offer it for trial or private use for free (for more details, see the license conditions). There are only few limitations to the phone. First of all, the audio subsystem will work only work with an acceptable quality if you are using

Hi, Is there is some form of VAD on * for SIP channels, cause I have a problem with MOH. I made an extension which simply plays MOH, when I dial that extension with my ATA188 MOH sounds choppy if I talk on the phone the MOH keeps playing. I saw the sip channel (show channel SIP/*) and I see no packets going in/out when I talk then packets shows going in/out. I don?t have this kind of problem

Hi, I'm using the new SipAddHeader application on Asterisk 1.2.1, here's a snip of my extensions: exten => _9XXXXXXX,1,SipAddHeader(P-Asserted-Identity: <sip:${CALLERIDNUM} exten => _9XXXXXXX,2,SipAddHeader(P-Asserted-Identity: tel:${CALLERIDNUM}) exten => _9XXXXXXX,3,Dial(SIP/${EXTEN}@${SIPTRUNK},,tT) exten => _9XXXXXXX,4,Congestion The problems is that Asterisk

Hi, Just downloaded the latest cvs from zaptel on my sparking new Athlon64 Centos4.2 system, but hitting a stumbling block... (sorry for the long post) #make cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -m64 -DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\"/etc/zaptel.conf\" -c -o gendigits.o gendigits.c cc -o gendigits gendigits.o -lm ./gendigits cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -m64

Hi, For MGCP users. Is there any success stories with any MGCP IAD vendor. I?m trying to find an IAD which works with Asterisk. I?ve tried the Cisco IAD 2430 without success; but SIP on this IAD works but it?s limited (no authentication, no notify messages, etc) and with higher density IAD (16 or more ports) it?s nice to control using MGCP. Any information will be apreciated ! Thanks. --

I have a cisco 7940 with the following sip.conf config: [Desk1.1] type=friend secret=****** defaultip=192.168.1.14 insecure=no mailbox=102 callerid="Desk1.1" qualify=500 canreinvite=no context=extensions host=dynamic group=2 I do not get message waiting indicator (mwi) on this phone. Is the another .conf file invilved in configuring this function other than the mailbox=xxx in the

I don't know if this is a problem with my cisco sip IP Phones or asterisk but I thought I would post here in case someone else has experienced this issue. When I make a call from my SIP cisco IP Phone to some remote IVRs I never get the rest of my soft keys, only the "End Call" soft key, and also DTMF doesn't work... its like the phone is acting like the remote end hasn't

Hi all I am looking at implementing asterisk at a company with two ISDN bricks (60 lines). I know that the VoIP will absorb at least on brick worth of lines but that still leaves me with a need for 30 ISDN lines. As far as I can tell most of the Digicom cards have 4 FXS ports and I've read on this list that at most two could coincide in a box simultaneously without causing an interupt

Hi Please help! I have one X101P and TDM400P in my asterisk Box When i make a call from * to PSTN, everything goes Ok, When the PSTN hangups or * hangups, the busy tone is detected and * disconnects the channel without problems. The problem occurs when the call comes from PSTN. When * hangups, the other end (at pstn) does not hangup, it only presents silence. Please tell me how to solve this

Hi all, A new version of DIAX (0.9.8a) is ready to be downloaded from the following locations: http://www.laser.com/dante or http://www.geocities.com/tdanro What's new in 0.9.8a: - unconditional autoanswer or based on CallerID (user configurable); - use any Ericsson/SonyEricsson GSM/PCS to control DIAX (feedback on the phone display) through Bluetooth (or serial cable). You do not even

Hello, I've been fighting one-way-audio issues with asterisk and SIP extensions for some time..., and I want to share with you my findings ;) My setup: * 1 ADSL router (Zyxel) * 1 Asterisk box with private IP, and interesting ports forwarded to it. * Several extensions, some local some remote The problem: * External extensions behind double nat don't get audio when they