similar to: No subject

device somewhere in your communication path, and since voice is picked up as DTMF, some device is also set to listen for inband DTMF. What is the origination source of incoming calls to your system? Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-07-08 4:24 PM, "das sandesh" <sandesh440 at gmail.com> wrote: Hi, We have few systems with asterisk 1.4.22.1 and we use sip

have problems with outgoing calls. When I tried this, the same way you did, I could make calles externally but had no audio each way reguardless of what I tried to pass to the sip provider. Best bet is to use what your sip provider can use or find another provider that that can do g722. That's what I did when I wanted to use g726. my2cents On Tue, Jun 29, 2010 at 2:42 PM, Mindaugas Kezys

supported, beside Idle, On call and Ringing ? Can we expect this list to match DEVICE_STATE's one (UNKNOWN | NOT_INUSE | INUSE | BUSY | INVALID | UNAVAILABLE | RINGING | RINGINUSE | ONHOLD) > Might be worth seeing if other phones do the same. > > S > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by

/var/lib/asterisk/sounds/soundfile.alaw /var/lib/asterisk/sounds/soundfile.wav to go from alaw to mp3, first convert to wav, then use lame <options> /var/lib/asterisk/sounds/soundfile.wav /var/lib/asterisk/sounds/soundfile.mp3 sox looks like it can ogg/vorbis, but mine doesn't list mp3. You might fetch the source for sox and see if it can do mp3; lame is probably just as easy to obtain

at least once a week I receive such an attack coming from a different ip. I will read the articles. Thanks again to everyone. Regards, Rodrigo Lang. 2010/6/29 Kenny Watson <kwatson at geniusgroupltd.com> > Hi, you can use fail2ban >

built-in; This doesn=92t matter because the moderator would have to use meetmeadmin or the confbridge equivalent to control the other functions. The moderator would either need two phones or a phone and a web = interface. Let=92s say Yves=92 =93special conference=94 is 5555. The moderator = would start using this command Exten =3D> s,1,meetme(5555) The participants would do Exten =3D>

Hey folks, I have a Digium TE205P working as a man in the middle: PRI line -------- Asterisk/TE205P -------- PBX The PBX is a Panasonic KX - TVP 100. Everything is working great except for one little issue. Asterisk isn't hanging up the PRI B channel when the PBX channel is hung up. I don't want to overload you with information but please ask if more is needed. I suspect I'm

sk hangup after respectively 899s 894s 898s<br> <br> In logs I see<br> <br> WARNING[8213] chan_sip.c: Retransmission timeout reached on transmission 52= 2eec628683-uy8xshd6wc21 for seqno 102 (Critical Request) -- See <a href=3D"= https://wiki.a" target=3D"_blank">https://wiki.a</a><br> Packet timed out after 6401ms with no response (or

, as it seems to be running Asterisk-11. =A0I&#39;ve previously installed A= sterisk-11+FreePBX in a VM, and this appears to be very similar. =A0Is ther= e any upside to using AsteriskNOW vs. Asterisk+FreePBX? Other than the obvi= ous fact that everything is nicely placed on an iso for ease of installatio= n?<br> <br> As for the actual upgrade, is it possible to step through each

<br> ;<br> ; Display certain channel variables every time a channel-oriented<br> ; event is emitted:<br> ;<br> ;channelvars =3D var1,var2,var3<br> <br> So if you want fu_callerid, set:<br> <br> channelvars =3D fu_callerid<br> <br> And, once that variable is set, you should get a NewExten event, you<br> should see the following

kernel-headers-2.6.32-279.el6.x86_64.rpm<br> <br> Alec Davis<br> <br> <br> --<br> _____________________________________________________________________<br> -- Bandwidth and Colocation Provided by <a href=3D"http://www.api-digital.c= om" target=3D"_blank">http://www.api-digital.com</a> --<br> New to Asterisk? Join us

to a fax (modulation and demodulation) over VoIP. Try again, the fact that you succeeded twice was pure luck, and as far as I understand that didn't even work out. Switch back to TDM. Your dial up modems want that magic thing called timing and no jitter that only TDM will give you. On Mon, Aug 29, 2011 at 2:56 PM, Aaron Krohn <akrohn at ewebforce.net> wrote: > We have an Ascend Max

faced this exact same problem a few times on more than one servers and it was 1) dialplan issue which was not hanging up the zap channels correctly 2) using more than 8 spans on a server. Asterisk can't handle more than 96 zap channels on T1s. FXO/FXS combinations can vary the number of spans but if you know what I mean by spans, in production don't use more than 6 spans. On 2010-03-17

with an active project, than a dead one. Otherwise who is going to patch vulnerabilities? Not me. I'm not a software developer. - Doug Mortensen Network Consultant Impala Networks P: 505.327.7300 . From: Steve Totaro [mailto:stotaro at totarotechnologies.com]=20 Sent: Thursday, March 24, 2011 11:11 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users]

crashing. So, as a first step to solving **that** problem, make sure asterisk is compiled with debug flags, dumps another core file, and then you do the "gdb asterisk <corefilename>", and get a stack trace. That should give us some idea of what happened. > > I have a fairly simple Followme sequence in place to see how it works > before I get into the complex scenarios.

/*! \brief Wait up to 16 seconds for first digit (FXO logic) */ static int = firstdigittimeout =3D 16000; /*! \brief How long to wait for following digits (FXO logic) */ static int = gendigittimeout =3D 8000; /*! \brief How long to wait for an extra digit, if there is an ambiguous ma= tch */ static int matchdigittimeout =3D 3000; -----Original Message----- From: asterisk-users-bounces at

/*! \brief Wait up to 16 seconds for first digit (FXO logic) */ static int = firstdigittimeout =3D 16000; /*! \brief How long to wait for following digits (FXO logic) */ static int = gendigittimeout =3D 8000; /*! \brief How long to wait for an extra digit, if there is an ambiguous ma= tch */ static int matchdigittimeout =3D 3000; -----Original Message----- From: asterisk-users-bounces at

/*! \brief Wait up to 16 seconds for first digit (FXO logic) */ static int = firstdigittimeout =3D 16000; /*! \brief How long to wait for following digits (FXO logic) */ static int = gendigittimeout =3D 8000; /*! \brief How long to wait for an extra digit, if there is an ambiguous ma= tch */ static int matchdigittimeout =3D 3000; -----Original Message----- From: asterisk-users-bounces at

/*! \brief Wait up to 16 seconds for first digit (FXO logic) */ static int = firstdigittimeout =3D 16000; /*! \brief How long to wait for following digits (FXO logic) */ static int = gendigittimeout =3D 8000; /*! \brief How long to wait for an extra digit, if there is an ambiguous ma= tch */ static int matchdigittimeout =3D 3000; -----Original Message----- From: asterisk-users-bounces at

/*! \brief Wait up to 16 seconds for first digit (FXO logic) */ static int = firstdigittimeout =3D 16000; /*! \brief How long to wait for following digits (FXO logic) */ static int = gendigittimeout =3D 8000; /*! \brief How long to wait for an extra digit, if there is an ambiguous ma= tch */ static int matchdigittimeout =3D 3000; -----Original Message----- From: asterisk-users-bounces at