similar to: transfer=mediaonly : can't hear nothing

Hi, The iax.conf is below and the trace. Any ideas please? disallow=all allow=g729 trunk=yes qualify=yes qualifysmoothing=yes nat=yes canreinvite=yes context=OutboundWS transfer=mediaonly Executing [082449627 at private:1] Dial("SIP/919-094d6e60", "IAX2/ECom-iax/2782449627|60|") in new stack -- Called ECom-iax/2782449627 -- Call accepted by xxx.xxx.xxx.x (format

Hi all Hmm, often when my Asterisk tryes to register, it get's the answer back: Got SIP response 302 "Moved temporarely" (and an IP). But it looks like it's not respecting this redirection and tryes again and again to register to the server configured in sip.conf instead of the one the SIP provider tryes to redirect to. Any known issues? Mit freundlichen Gr?ssen Benoit

Something I've just noticed that might persuade me to move to 1.4 ... in iax.conf, there is a new option: transfer=mediaonly Does this mean that it keeps itself in the loop as far as signalling/CDR is concerned, but lets the media stream go between the 2 endpoints? ie. Asterisk A <-> Asterisk B <-> Asterisk C Where B keeps track of CDRs, so right now I have

Hi, I compressed many files using bzip2, but in some cases it crashes. I believe that it is related to dovecot indexes. How to reproduce: Inside a Maildir folder without any indexes yet, compress a message with bzip2 and tryes to access it. At my server is showed the error: Jun 24 00:18:20 maildev dovecot: IMAP(xxx at xxx): FETCH for mailbox Trash UID 1 failed to read message input: No such

Hi! In October I worked with a client-server (using winsock.dll) 16 bit aplication emulated in wine (Not using ODBC), it worked fine. This aplication needs to connect to a server using port ctsql 5557/tcp, this service is in my /etc/services. And wine made all realy fine (Version 20011108). With the release 20011226 (I think)started the problem. Something changed with winsock 16-bit. I've

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 all, i have the following setup PSTN -> routing server (asterisk 1.6.2.11) -> IAX -> callcenter asterisk 1.6.2.9 -> SIP -> agent Does work quit fine - then agent does have the abibility to transfer a call to a third party - the agent can initiate the transfer over a web interface - it does generate a asterisk manager atxfer request... So agent does initiate transfer - call

hi.. im using ldap. what happens when i add a new ldap user and that user tryes to login to dovecot ? does dovecot create the mailbox on first login? or do i have to send a mail first ? thx

I new in asterisk world so, please, forgive me if I say something stupid. At least, and after a lot of tryes, the isdn card seems to be registered: [chan_zap.so] => (Zapata Telephony w/PRI) == Parsing '/etc/asterisk/zapata.conf': Found -- Registered channel 1, PRI Signalling signalling -- Registered channel 2, PRI Signalling signalling -- Automatically generated pseudo

Hello samba experts, I want to configure samba with mysql as backend. I searched on google and i found some info about this subject but it seems to be not really what i nedd. I said this because i found few howtos about how to configure samba to read password from mysql database to authenticate users. Is good but not enough. I don't want shell users with different password for samba!!! I

After updating to the latest CVS Feb 17 15:20:03 WARNING[15317]: config.c:819 read_config_maps: The 'sipfriends' table is obsolete, update your config to use sipusers and sippeers, though they can point to the same table. == Binding sipusers to mysql/asterisk/sip == Binding sippeers to mysql/asterisk/sip Feb 17 15:20:03 WARNING[15317]: config.c:823 read_config_maps: The

I can't create new folder in a czech Win95 explorer on a unix using samba 1.9.18p7. The problem is that Windows exporer uses for the creation new folder a name containing characters above 128. The explorer creates new folder in a two phases: the first It creates directory named Nov<A0> slo<A7>zka and next it gives to user posibility to change this name. Crazy !!? The first phase

Hi, As mentioned we have the problem that sometimes (could be up to a view times a day) for the calling party (SIP Device) you here ringing. The called party however answered the phone, but hears nothing. The calling party keeps ringing until the phone is hangup. First I thought maybe the card or the server has a problem, so I changed from a PCI beronet 4bri to a Junghanns 4bri PCIexpress and

Hi All, I connect SIP phones and H323 phones, with *. But, if I call from H323 to SIP phone, i can hear nothing. I'm using h323 library. I found some e-mails at the Digium's Mail-list with the same problem. But, I coudn?t find the solution yet. Could you please help me?, any idea?. It's quite urgent. Thanks. -------------- next part -------------- An HTML attachment was

Hi all, I have two problems with h.323 in * The first one is, I can call my voip-phone, (exten => 59305004,1,Dial(H323/${EXTEN}@192.168.0.41)) BUT, I hear nothing in h.323 debug mode: *CLI> Allowed Codecs: Table: GSM-06.10{sw} <1> Set: 0: 0: GSM-06.10{sw} <1> -- Making call to 59305004@192.168.0.41. == New H.323 Connection created.

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 think I have 4 options. 1, Modify chan_sip.c to update a new field in sipusers realtime table with the status of the sip peer/user. Then use agi to dial sip calls. Check the status field if OK then dial the fullcontact from the sip table. If not goto voicemail or where ever else I want the call to go.. The UA would only register to one server, so only one server *should* be writing to the

plug in your 302 USB enter: lsusb you should see: Bus 001 Device 005: ID 08bb:2902 Texas Instruments Japan PCM2902 Audio Codec look at syslog: ... usb 1-1.2: new full-speed USB device number 5 using dwc_otg usb 1-1.2: New USB device found, idVendor=08bb, idProduct=2902 usb 1-1.2: New USB device strings: Mfr=1, Product=2, SerialNumber=0 usb 1-1.2: Product: USB Audio CODEC usb 1-1.2:

Hi. Finally I got icecast and darkice to work. I can connect, but now I've got a problem. I can't hear anything. I can hear sound from the computer, but not from Another computer when connecting. I've got an usb mixer. behringer xenyx 302. I'm using alsa. What can this be? /Kristoffer -- Kristoffer Gustafsson Sal?ngsgatan 7a tel:033-12 60 93 mobil: 0730-500934

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)) <-