similar to: FWD problem

============= SJphone Log ============ Outgoing SIP session Respondent: (sip:8612@192.168.2.2) Remote client: Started: May 26 16:33 Accepted: no Ended: May 26 16:34 End reason: Call rejected: 503 Service Unavailable =============== Asterisk Debug ================ Executing Dial("SIP/2201-a83e", "IAX2/<FWDNUMBER>:@iax2.fwdnet.net/612|60|r") in new stack --

When I try to call out to FWD over IAX2 I get: Call rejected by 65.39.205.121: Unable to negotiate codec I'm using asterisk-1.0.5 (the same settings works fine with *0.9) I've standard settings in iax.conf [general] bindport=4569 register => xxxxx:xxxxxx@iax2.fwdnet.net [iaxfwd] type=user context=fromiaxfwd auth=rsa inkeys=freeworlddialup disallow=all allow=ulaw -- #Joseph

I have looked all over the site(s) for help. But heres the problem. Im missing something. In coming works fine from FreeWorld via IAX. But when Dialing out i get: May 13 13:42:01 WARNING[1150495040]: chan_iax2.c:5256 socket_read: I don't know how to authenticate iaxtel to 65.39.205.121 my IAX.conf if as follows [general] port=5036 register => ######:xxxxxxxxxxxxx@iax2.fwdnet.net

> Well that didn't work....I now get this error > > > Jan 12 16:56:21 NOTICE[4989]: app_dial.c:746 dial_exec: Unable to > create > channel of type 'SIP' > == Everyone is busy/congested at this time > -- Executing VoiceMail("IAX2/iaxfwd@65.39.205.121:4569/5", "b") in > new > stackJan 12 16:56:21 WARNING[4989]:

Hi ALL Have installed asterisk@home 1.0 On FWD DID's, appears that 2 calls are generated to the inbound extention. I have confirmed this on a number of friends boxes also. Does anyone have a fix for this ? I set the DID simply to a custom context and it did the same... Anyone have a way to fix this ? Here is the output...... -- Accepting AUTHENTICATED call from 65.39.205.121, requested

I've searched WIKI and Archives but nothing. Im getting: -- Called username:password@iaxtel.com/1800somenumber@iaxtel Jun 21 17:04:12 WARNING[1158883520]: chan_iax2.c:5097 socket_read: Call rejected by 69.73.19.178: Unable to negotiate codec -- Hungup 'IAX2[Iaxtel]/8' == No one is available to answer at this time -- Executing Hangup("SIP/104-b8eb", "")

Hi, I got my * working fine with FWD at office with 2 extensions, i receive calls and i can make calls thru FWD. I got also my * at home, and i connected it using auth=rsa. From my home, i can make calls using my office iax, but if i try to redirect incomming calls from FWD to my * at home, it rejects the call. I created the pub/key pairs for rsa and its working ok and i just pasted the

Hi, testing a new fwd account, dialling from sip4030 to my FWD number, sip4021 rings as defined in extensions conf. Why is this happening twice? -- Executing SetCallerID("SIP/4030-a7f2", ""HTCAS"") in new stack -- Executing Dial("SIP/4030-a7f2", "IAX2/617533:xxxxxx@iax2.fwdnet.net/617533|60|r") in new stack -- Called

Hi Folks, Just wondering if anyone can give me some pointers, I'm configuring Asterisk to talk to FWD's new IAX service. The asterisk server is behind an iptables NAT Firewall, with port 5036 forwarded: $IPTABLES -t nat -A PREROUTING -p udp -d $EXTERNAL_IP --dport 5036 -j DNAT --to-destination 172.16.20.200:5036 I can make outgoing calls just fine, but when I receive an inbound call

Hello all, Here is my problem, I try to place a call to FWD (free world dialup) trough my asterisk PBX. my config is as follow: extensions.conf ---------------- [internal] exten => 613,1,Dial(IAX2/iaxfwd-outbound/613) (service echo de FWD) exten => xxxxxx,1,Dial(IAX2/iaxfwd-outbound/xxxxxx) mon numero FWD exten => yyyyyy,1,Dial(IAX2/iaxfwd-outbound/yyyyyy) celui d'un ami FWD

Hello, I have 2 channels in iax.conf [iaxfwd] type=user callerid= Free World Dialup inkeys=freeworlddialup auth=rsa context=incoming qualify=yes [iaxfwd-outbound] type=peer host=iax2.fwdnet.net username=xxxxxx secret=*********** auth=md5 The problem is: When I tell FWD to call me I have this output in my asterisk consol: Executing Dial("IAX2/iaxfwd-outbound-3",

I have a user, running CVS a/o 11/23/03, who has complained about "phantom" messages showing up days or even weeks after she has deleted them. So I asked her to let me know when it happened again, and she called a few minutes ago. The directory listing below shows a listing of the /var/spool/asterisk/voicemail/default/XXXX/Old directory, and to my surprise the messages are indeed

Hi, I'm having trouble getting asterisk to report MWI to a Cisco CCME. I record a message in mailbox 29, but the subsequent MWI notifications I see continue to report no messages waiting. Are they reporting for the wrong mailbox? Is there some other option I have to set or change? I'm running asterisk-1.4.22 Since the mailbox is in [home] in voicemail.conf, I've tried things like

Does anyone have a quick and dirty script for defragmenting mailboxes? i.e.: -rwx------ 1 root root 80553 Oct 20 16:27 msg0000.gsm -rw-r--r-- 1 root root 218 Oct 20 16:27 msg0000.txt -rwx------ 1 root root 781164 Oct 20 16:27 msg0000.wav -rwx------ 1 root root 79360 Oct 20 16:27 msg0000.WAV -rwx------ 1 root root 7260 Oct

When somebody call me on my pstn # cable connected to my fxo card it does not work when I check my computer the following error shows Connected to Asterisk CVS-v1-0-12/05/04-19:46:25 currently running on asterisk1 (pid = 2160) Verbosity is atleast 3 -- Remote UNIX connection -- Starting simple switch on 'Zap/1-1' == Starting Zap/1-1 at incoming,s,1 failed so falling

Hi, We can't read the messages in our mailbox always getting -- <SIP/tootaiAUDIO-00000001> Playing '/var/spool/asterisk/voicemail/default/100/Old/msg0002.slin' (language 'fr') [Oct 11 13:24:50] WARNING[26778]: app_voicemail.c:7802 play_message: Playback of message /var/spool/asterisk/voicemail/default/100/Old/msg0002 failed As you see Asterisk try to read

Hello everybody, I was trying to install a web interface to my Voice Mail, Vmail.cgi I can log on it, list messages, but no play with the following error msg; "Hrm, can't seem to open /var/spool/asterisk/voicemail/default/234/INBOX/msg0001.WAV" Remark: playing the message msg0001.WAV directly OK Any smart guy up there could help ? Thanks, --------------------------------- Do

I am trying to set up a simple Asterisk server. All it's going to do for now is to act as my voicemail box. I've got a DID from Voicepulse, and am using IAX (I'll get to SIP someday when I want to circumvent the phone company for long-distance, but for now I'd be happy to get a trial version of Asterisk running). So far, I've managed to set up voicemail.conf, extensions.conf

Hi, I am fairly new to asterisk. I am currently testing my first setup. I've been able to debug most of the problems to make asterisk work with my hardware setup until this time. Currently I have the following issue: Voicemail is running but when I test to leave a voicemail thru my incoming PSTN channel (voicetronix / vpb), asterisk will not detect sound (according to the log) on that

Which version? Similar issues parsing callback number in 1.2.12 > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of Kristian Kielhofner > Sent: Thursday, September 28, 2006 10:27 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [asterisk-users] Voicemail callback