similar to: chan_iax2.c: No more space

Hi! Is it somehow possible to evaluate the SIP response code inside the dialplan? I have an Asterisk server which forwards requests to various PSTN gateways with SIP. If the Dial() attempt is not successful I want to differ at least these 3 options: - called destination is busy (486): e.g. activate auto-redial - called destination does not exist, unassigned number (404) - gateway is broken,

Hi, As of v 1.6.1.1, can anyone tell me what the current possible values for DIALSTATUS could be? I found http://www.voip-info.org/wiki/view/Asterisk+variable+DIALSTATUS but believe it is outdated since there is no FAIL or FAILED in this list. Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I have remove the comment defor res_odbc.so and res_config_odbc.so in my modules.conf, but the module is still not loading when I do: module show like odbc I have o module returned anybody knows why? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090127/0963b5a4/attachment.htm

Hi! http://www.voip-info.org/tiki-index.php?page=Asterisk+sound+files+international#German lists a plenty of sound files for German. Can someone recommend one for Asterisk 1.4 (any maybe 1.6 soon). thanks klaus

Hello, I am trying to do : Exten =>_X.,n,Dial(SIP/ser_sei0/1130,L(10208400:61000:10000)) But it return that error: [Jun 2 10:04:44] WARNING[18920]: app_dial.c:1623 dial_exec_full: Invalid timeout specified: 'L(10208400:61000:10000)' Why? I forgot something ? Thank you Cordialement, BERGANZ Fran?ois P Pensez ? l'Environnement, n'imprimez ce mail que

Dear, I added new field to cdr table , named "service" and type varchar(20), but in extensions.conf with the following command, nothing to be saved. exten => _X.,1,Set(CDR(service)=OUT) does asterisk support this ability ? is any setting must be changed, before that ? best Mani

Does anybody think it would make sense for /etc/init.d/asterisk to run /etc/asterisk/startup.d/*.sh on start like safe_asterisk did? Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied -> http://www.amooma.de Gesch?ftsf?hrer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP

On starting Asterisk (1.4) I get a whole bunch of WARNING[5858]: pbx_ael.c:4040 ast_compile_ael2: Warning: file /etc/asterisk/extensions.ael, line 932-932: Empty Extension! I find it a bit disturbing that this message has a level of WARNING (instead of NOTICE maybe) because the extensions in question are empty on purpose. The only reason they exist are the hints. hint(SIP/3000) 3000 => {}

Is there any method in Asterisk to enable the updating process into another SQL database through entering IVR options during the call. Thanks a lot. _________________________________________________________________ Invite your mail contacts to join your friends list with Windows Live Spaces. It's easy!

Hi all, I would like to ask please if there is a way to play a ring back tone from asterisk when the customer try to make a call...I already added the ringing function to the context in extensions .conf and it work perfectly...But the issue that the asterisk server is stoping playing back his own ring back tone as soon as it detect a ring back tone coming from the carrier side... Is there a way

Hello Asterisk People, I am having trouble connecting asterisk to a huawei SoftX3000 Switch, i can succesfully connect other softphones like Zoiper, but when it comes to Asterisk SIP Client, the system doesn't authenticate, i have the following configuration: peer: 10.220.0.2 username: 4857768 password: 4857768 the configuration is as follows: in the general section: register =>

Hi, is it possible to hangup a channel from another channel? I want to finish a call from another channel, but if I put exten => h,n,HangUp(channelname) it doesn't hangup... Is that correct? Thanks, _________________________________________________________________ -------------- next part -------------- An HTML attachment was scrubbed... URL:

I would like to know if I can set Call Forwarding on an extension but allow direct calls from the extension it is forwarded to. Example: Extension 100 sets call forwarding (all) to extension 101. All calls to 100 are immediately forwarded to 101 as expected. However, if 101 tries to transfer a call to 100 or tries to call 100 directly, it sounds "busy" because it obviously goes into

I just want to confirm but it seems that if-then-else is not permitted in case structure. It was not really documented but it seems to be the case. Can anyone confirm? switch(${DIALSTATUS}) { case NOANSWER: { // if-then-else not permitted If (${ael-var} = 1) { Playback(beep);

Hello, I'm just wondering if anyone has fixed the 'InUse&Ringing' problem. * v1.4.23.1 Ta

Hi My asterisk output is: chan_sip.so => (Session Initiation Protocol (SIP)) Asterisk Ready. -- Registered SIP '201' at 192.168.0.55 port 33906 -- Saved useragent "X-Lite release 1011s stamp 41150" for peer 201 -- Executing [907768385144 at default:1] Dial("SIP/201-083e75c0", "ZAP/g1/907768385144|60") in new stack [Oct 4 11:54:27]

~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ CLI Output : ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ vicidialnow*CLI> == Parsing '/etc/asterisk/manager.conf': Found == Manager 'sendcron' logged on from 127.0.0.1 == Parsing '/etc/asterisk/manager.conf': Found == Manager 'sendcron' logged on from 127.0.0.1 == Manager 'sendcron' logged off from

Hi all, I have a question: how can I see hints of a remote Asterisk in IAX2 trunk?? I want to set BLF on my phones to look state of other phones also in other Asterisk server. Someone have any idea or solution? I use Asterisk 1.4.24. Thanks all Marco -------------- next part -------------- An HTML attachment was scrubbed... URL:

I setup two asterisk servers with identical settings (same extensions, same queues, etc). Each one is connected to the same amount of incoming/outgoing links (1 PRI, 4 BRI, 1 IAX friend, etc, on each box). Most extensions are sip and they register via DNS SRV and other methods so that the two servers are load balanced. Incoming PSTN calls (BRI) reach 50% each server so that's load balanced

I have an account with FWD and I have configured my SIP.conf with [fwd] type=friend secret=password username=901835 host=fwd.pulver.com But when I am trying to dial out my own DID , I dont see any call landing in asterisk. In extension.conf (vicidial) file I have exten => 2062036895 ,1,Ringing() exten => 2062036895 ,2,Wait(1) exten => 2062036895 ,3,Answer() exten => 2062036895