similar to: 302 redirection issue

I've been googling and talking with Libretel about my setup and the fact that incoming calls to my asterisk box through the Libretel number reach my box (I hear the greeting being played) but then don't accept DTMF. Here is a rough diagram of my setup: Asterisk | server | NAT <------------ Libretel | router | Note that there are NO SIP

Hi, I have the following problem that when asterisk receives SIP response 302 it cannot forward the call I get such debug: [Jan 4 10:43:27] WARNING[18671]: channel.c:3281 ast_request: No channel type registered for 'Local' [Jan 4 10:43:27] NOTICE[18671]: app_dial.c:505 wait_for_answer: Unable to create local channel for call forward to 'Local/poczta at routing-sip' (cause = 66)

Hi have set the following tbf tc qdisc add dev eth0 root tbf rate 0.5mbit \ burst 5kb latency 70ms peakrate 1mbit \ minburst 1540 I want to add a filter so the ip traffic pass from it.. plz help me __________________________________ Yahoo! Mail Stay connected, organized, and protected. Take the tour: http://tour.mail.yahoo.com/mailtour.html

Hi Friends, I have a problem with my Wine Installation in FC2-Athlon. I was following the PDF File - Configuring Wine Chapter 5 and it says that the "config" file is in ~/.wine/config. I can find the "config" file in the /etc/wine/config. Can you enlighten me on this? Actually, I can run already the "wine winmine.exe" and "wine notepad.exe" Also, I was

After buying some additional lines from my telco, I recently had my phone vendor wire the additional lines from my phone box into an amphenol connector that's plugged into my channel bank (Adit 600). However, although I make the following changes in my zaptel.conf and my zapata.conf files respectively and then reboot, the new channels, when selected for an outbound call, resume a dialtone

Probably doesn't help diagnose the problem.... but there were also audio problems experienced with this cvs version even on LAN / sip2sip / no transcoding > > ERROR[1171] UTILS.C:509 TVFIX: WARNING NEGATIVE TIMESTAMP -194931. ... > > I will be looking into this issue later today. __________________________________ Yahoo! Mail Stay connected, organized, and protected. Take

Hi: I need help setting-up multiple calling cards with different prices for the same routes using astcc. All my calling cards' routes now have the same price, but I need to be able to set multiple calling cards with different prices for the same route. I appreciate your feedback of How I can do that. Thanks; Chawki __________________________________ Yahoo! Mail Stay

Hi all, i am running a application which uses ODBC DSN. this is file and connecting with MS sql server. but when i run my program as $ wine c:\\crm\\PMonitor.exe i get fixme:vxd:VXD_Open Unknown/unsupported VxD L"secprov.vxd". Try setting Windows version to 'nt40' or 'win31'. But setting to this versions doesnot help as it provides dsn error like nddeapi.dll could not

--- Dan Pritts <danno@umich.edu> wrote: > Hi all - > > I've just finished building flac in the "darwinports" environment > on MacOS 10.3.9. > > The port maintainer (i've cc'd him) had disabled the xmms plugin > build. > > I wanted that, so I changed the portfile and built locally, yada > yada. > > I've run into three problems,

> All TCP connections have a port number, the default > for http is 80. Thanks. Good point. I was wondering how to handle this in icecast.xml. > As for the source of the relay, it can be shoutcast > or icecast, because > the relay is classed as a listener by the source. What if the source is neither icecast or shoutcast - but another mp3 stream source? If it matters, why?

Hi there, im having some troubles with my asterisk service, sometimes when im trying to make an outbound call, to any of the phones configured on the asterisk box, it enters inmediatly to voicemail and then hungs up. After that its necessary to stop the service and putting up again manually. Here is a piece of my log file when a call is trying to incoming: "Jun 9 06:30:16

Hello, I have a question . dose asterisk supports "302 Redirecting..." ? I have SIP Server "Not Asterisk" and my Asterisk is registering as a client for this device . when i try to call another client registered to the same SIP server i got Busy Tone and here is the asterisk CLI output ----------------- -- Got SIP response 302 "Redirecting..." back

Hiji wrote: >--- "David F. Colwell" <dfcolwell@dfcolwell.com> >wrote: > > >>Hiji et al, >> >>Couldn't find Half-Life or Counter-Strike in the DB >>yet they returned... >> >>Submitted version rejected >> >> >> >------------------------------------------------------- > > >>The version you

Hi all, I have installed a test machine with asterisk in order to try it. I have a problem with capi channel (chan_capi 0.3.4a). When an external call directed to an internal Ip phone is not answered I obtain this warning repeated many times: .... .... Jul 12 16:13:43 WARNING[1209214400]: app_dial.c:302 wait_for_answer: Unable to forward frame Jul 12 16:13:43 WARNING[1209214400]: app_dial.c:302

Hi folks, this does look like a bug to me: Asterisk replaces the @63.214.186.6 by @context which obviously leads to a failure. Any comments, do I have a configuration issue on my side that I missed? Cheers, Philipp -- Executing Dial("SIP/philipp-bd5f", "SIP/992365264680@nikotel- out|90") in new stack -- Called 99xxxxxxxxxx@nikotel-out -- Got SIP response 302

We have a situation where we get no early media in this call flow: VoIP origination provider Server1 (our server) Customer server Customer phone with call-forward set Server1 to dial the forward-to number Then there is no early media while the forward-to number is ringing. Our server is Asterisk 1.6 and theirs is 1.8. I tried promiscredir=yes and then the calls fail altogether because rather

Hey, I'm having some errors whenever I dial out and I can't dial in at all. I'm using NuFone as my provider just so you know. Jan 11 17:39:46 WARNING[1771]: chan_oss.c:413 soundcard_setinput: Unable to re-open DSP device: No such device Jan 11 17:39:46 WARNING[1771]: chan_oss.c:572 oss_write: Unable to set device to input mode Jan 11 17:39:46 WARNING[1771]: app_dial.c:359

Hi, I've got an important question: I use an E100P directly connected to PSTN, but it does not *really* work as it should be: exten => 1000,1,Dial(Zap/1/1234) BUT: It does NOT dial "1234" but it says in debug mode: -- Called 1/72976451 Apr 26 00:53:00 WARNING[10251]: chan_zap.c:5979 zt_pri_error: PRI: !! Facility message shorter than 14 bytes -- Channel 1, span 1 got

Hi, Updating zaptel gives me this during the make. Any ideas, the searches and Wiki gives me no hints. In file included from /usr/src/linux-2.4/include/linux/fs.h:19, from /usr/src/linux-2.4/include/linux/capability.h:17, from /usr/src/linux-2.4/include/linux/binfmts.h:5, from /usr/src/linux-2.4/include/linux/sched.h:9, from