Re: [asterisk-users] Turn on SIP debugging from DialPlan

Hi, If you are ok with starting debug via external system call, why not to use something like this (I used to use something similar, it worked): exten => _XXX,1,System(/usr/sbin/asterisk -rx ‘sip set debug peer PEER’) same => n,Set(debug_on=1) same => n,Dial(SIP/PEER/${EXTEN}) exten =>

Re: [asterisk-users] Call transfer problem.

You have to use attendant transfer, not blind. - A calls B - B answers on line 1 (button 1) - B has to use line 2 (push button 2) to call C, C sees call coming from B, the same does asterisk - while having line 2 active, he pushes button transfer followed by button line 1 - A speaks with C On

Re: [asterisk-users] 1.4.22 vs 1.4.21.2 - IAX2 regression ?

Just FYI, this issue has been fixed. On Tue, Nov 4, 2008 at 3:12 PM, Igor Zamocky [EMAIL PROTECTED] wrote: http://bugs.digium.com/view.php?id=13645 Thanks Igor, we'll keep an eye on it. -- exvito ___ -- Bandwidth and Colocation Provided

Re: [asterisk-users] Is SIPPEER curcalls working for you ?

Title: Re: [asterisk-users] Is SIPPEER curcalls working for you ? Did You triedhttp://www.voip-info.org/wiki/view/Asterisk+sip+limitonpeers? Hi, In this threadhttp://lists.digium.com/pipermail/asterisk-users/2008-October/219592.html, I wondered whether SIPPEER curcalls was working. I

Re: [asterisk-users] 1.4.22 vs 1.4.21.2 - IAX2 regression ?

http://bugs.digium.com/view.php?id=13645 Hi list, I just experienced an odd behaviour in 1.4.22 vs 1.4.21.2. To cut a long story short, IAX2 is not tx-ing hangup... Scenario is composed of two asterisk systems A and B. A receives calls from IAX users X, Y, Z, etc, does some validation

Re: [asterisk-users] BRI or PRI callerid

And do You have usecallingpres=yes in your zapata.conf ? Hi,I try to get anonymous calling working on ZAP. But I am unsuccessful onPRI as well as on BRI. I tried all parameters from the application SetCallerPres(). Nothingworked. I even traced with my ISP and they told me that I am not

Re: [asterisk-users] Pre-paid Billing

Isn't 'don't allow multiple calls' acceptable solution? At least, it's the simplest one :) I can imagine solution with multiple calls allowed, but it needs some external synchronous processing. With every call you should start process, that will decrement user's balance based on dialled

[asterisk-users] MGCP Thomson, early transmit problem

Hello, I've got strange problems trying to run asterisk with MGCP ip phones (Thomson ST2030). Situation: user A - pstn --- ASTERISK - mgcp -- user B User A, connected behind a PSTN, tries to call User B. After dialing User B's number, call comes to ASTERISK,

Re: [Asterisk-Users] asterisk still tries native bridging

:-(( There's nobody with any idea here? :-((. I need to force * to not try native bridging, at least when there are different codecs used. In current config * tried native bridge, it fails, but CDR has been already generated and writed :-((. Thanks a lot for your time (and

[Asterisk-Users] asterisk still tries native bridging

Hello, I've problems with following - - --- --- PSTN | --- isdn --- | A | - iax2 -- | B | - --- --- On [B], there is unconditional call forwarding set back via [A] (dialparties.agi is used)

Re[2]: [Asterisk-Users] Help with Queue Cmd

Try to add n into option .. http://www.voip-info.org/wiki-Asterisk+cmd+Queue ... 'n' no retries on the timeout; will exit this application and go to the next step. ... dzajro HengWee Chin wrote: Hi, I have a problem with the queue cmd. I am trying to redirect an incoming

Re: [Asterisk-Users] Cisco ATA and G729

Hi * However, when i set my Cisco ATA to G711, i can't hear any sound unless I press at least two or three keys(any random keys). I am using the demo context of extension.conf file. Can that be due to a fast start problem? Anyone knows how to checkthe faststartcapabilities of an ATA 186?