Allowing Calls - Maybe I'm Just Stupid...

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Hi again!

About my previous E-Mail...

I though about it and I think, that maybe I'm just very stupid... Since I called an INTERNAL number, Asterisk tried to call it.

I tried right now to call an EXTERNAL number (using my context [myproxy]) and the behavior is NOT the same... Not 100% correct, but it tries the right way...

Now my problem is to check in my dialplan if the peer, that originate the call, is reachable, and if not, to give an error...

Is there any function to know if the peer is reachable?

Thanks Luca Bertoncello (lucabert@lucabert.de)

Asterisk Users 3 months ago 6 Answers

Call Accepted From Not Registered Peers?

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Hi list!

So, new day, new problem...

I tried right now to call from my cellphone a peer in my Asterisk. The cellphone has correct credentials, but it's NOT registered on my Asterisk, now.

I just tried to call a peer in my network, from a peer not yet registered. And it works... :(

The very curious thing is, that I can't find how the call will be accepted... Every section in my dialplan has a log, and no log will be displayed on the CLI...

I just see:

== Using SIP RTP CoS mark 5 -- Executing [00493511111111@default:1] Dial("SIP/00491773333333-0000000b", "SIP/00493511111111&DAHDI/1") in new stack ==…

Asterisk Users 3 months ago 1 Answer

Chan_mobile And Hardphones?

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Hi,

I have configured a certified asterisk 13 server with chan_mobile and res_pjsip. I have a Cisco 7940 hardphone and I use ekiga as softphone client.

Now the problem is, using the hardphone I'm able to call the softphone and hear everything properly. But when I call from the hardphone to some number that has to be dialed via chan_mobile, I'm not able to hear what the other side says (I get some noise). Though whatever I say on the SIP phone is audible on the other side.

Now when I use ekiga with the *same* codec (ulaw/alaw), I'm able to hear both…

Asterisk Users 3 months ago 1 Answer

Problem Asterisk Voicemail Message Records

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Hello!

I've got a little problem with Asterisk (11.14.1), the voicemessages are kinda limited to 40 seconds (average) aproximately; because when a message reach this long I got a cut in the file (*.wav) after I got this message:

WARNING[15035][C-000021ef]: format_wav_gsm.c:418 wav_read: Short read (20) (Resource temporarily unavailable)!

Does anyone got this problem, any idea of what is happening?

Thanks

Asterisk Users 3 months ago 1 Answer

Connecting Two Asterisk

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Hi again!

I always try to get my mobile phone work with my Asterisk. I tried to install Asterisk on my PC (with public IP), but it has problems, too... I think, my UMTS-Provider doesn't want to connect to dynamic IP or my DSL-Provider does not want it, too, since I have no problem to connect and get a very good audio quality if I connect to other SIP-Provider or to an Asterisk (SAME configuration!!) installed on my Server...

Well, I will try to configure the Asterisk on my Server to act as "proxy" so that all phones at home talk with…

Asterisk Users 3 months ago 4 Answers

Problem With NAT - Part 2

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Hi again!

I decided, "just for fun", to install Asterisk on a server of mine (available in Internet) and to log on my mobile phone on this server.

This Server communicate with my Asterisk at home and if I call a phone at home from my mobile phone (logged on the Asterisk on the second server), it work perfectly with a very good audio quality.

If I log my mobile phone (and just tried with the PC of my father) direct on my Asterisk at home it does NOT work or the audio quality is very poor...

So, I'm thinking I have somewhat misconfigured…

Asterisk Users 3 months ago 0 Answers

Curious Problem With NAT

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Hi list!

Since the internal calls work as expected and I can register my Asterisk on an external provider, I'd like to add a new feature and allow my mobile phone to connect to my Asterisk and manage calls.

Well, first of all, my Asterisk is NOT direct on Internet available, but behind a NAT. So I configured my sip.conf:

localnet2.168.200.0/24 externhost=myhost.noip.com externrefresh0

Then I added the peer in my users.con:

[00491771111111] fullname = 00491771111111 secret = MYVERYSECRET type=peer nat=yes qualify=yes qualifyfreq` hassip = yes dahdichan = 1 transport=udp,tcp callwaiting = no context = default host = dynamic dtmfmode=rfc2833 dial=SIP/00491771111111

and finally "core reload".

On my Gateway…

Asterisk Users 3 months ago 11 Answers

11.17.1 : ReceiveFax Then Signal 11 ??

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dialplan

[FaxIncoming]

exten=s,1,NoOp(Incoming fax on 46-va) same=n,Set(FAXFILE=/var/spool/asterisk/fax/${STRFTIME(${EPOCH},,%Y%m%d)}_${STRFTIME(${EPOCH},,%H%M)}) same=n,Answer() same=n,ReceiveFAX(${FAXFILE}.tif,df) same=n,Hangup() exten=>h,1,NoOp(FAXSTATUS: ${FAXSTATUS} FAXERROR: ${FAXERROR} FAXPAGES: ${FAXPAGES} @ bitrate ${FAXBITRATE}) same=n,System(scp ${FAXFILE}.tif asterisk@asterisk:/home/asterisk/ec2fax/ ) same=n,System(ssh asterisk@asterisk "/home/asterisk/bin/email-ec2fax.sh ${FAXFILE} ${CALLERID(name)} ${CALLERID(num) " )

cli ......... -- Executing [s@FaxIncoming:3] Answer("SIP/ec2faxcall17-00000000", "") in new stack -- Executing [s@FaxIncoming:4] ReceiveFAX("SIP/ec2faxcall17-00000000", "/var/spool/asterisk/fax/20150605_2137.tif,df") in new stack -- Channel 'SIP/ec2faxcall17-00000000' receiving FAX '/var/spool/asterisk/fax/20150605_2137.tif' ip-172-31-53-29*CLI> Disconnected from Asterisk server Asterisk cleanly ending (0). Executing last minute cleanups

syslog

asterisk[11540]: -- Executing [s@FaxIncoming:3] Answer("SIP/ec2faxcall17-00000000", "") in new stack asterisk[11540]: -- Executing [s@FaxIncoming:4] ReceiveFAX("SIP/ec2faxcall17-00000000", "/var/spool/a...ew stack asterisk[11540]: -- Channel 'SIP/ec2faxcall17-00000000' receiving FAX '/var/spool/asterisk/fax/20150605_2137.tif' systemd[1]: asterisk.service: main process exited, code=killed, status/SEGV asterisk[11587]: Unable to connect to remote…

Asterisk Users 3 months ago 0 Answers

Logging In "local Time"

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Hi again!

I just noticed, that my Asterisk (running on an OpenWRT-Switch) writes the logs using GMT... On the Switch the time is right configured and a "date" says me the current LOCAL time.

I didn't found in logger.conf or other file an option to set the timezone. Can someone help me?

Thanks Luca Bertoncello (lucabert@lucabert.de)

Asterisk Users 3 months ago 1 Answer

Missed Call

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Hi list!

I configured Asterisk to forward the incoming call for a number to both phones. I wrote that:

exten => _00493512222222,n,Dial(SIP/00493512222222&SIP/00493511111111,,R)

of course it works... Now the problem is, that when a phone get the call, on the other phone I get "1 missed call"... Is it possible to avoid that and signaling the other phone, that the call was not "missed"?

Thanks a lot Luca Bertoncello (lucabert@lucabert.de)

Asterisk Users 3 months ago 1 Answer