similar to: Asterisk PRI problems (Crashing when full)

We have recently started routing about 3 PRI's worth of traffic thru our asterisk box. The text on the console now flys by so damn fast, I can't really see what the heck is going on. Even with verbosity 0 and debug 0 it is still so fast. Is there some way I can attach to the console in a way that will allow me to grep or otherwise filter the text so I can focus on something in

Tie line is type em_w (old school stuff) to TE110P. Phone rings on the asterisk side, but the calling party does not hear the ring through sound. If I pick it up within the first two rings it goes through and I can talk otherwise our old switch drops the call. Anyhow...here is my config if anyone can shed some light on it. It used to work with HEAD a few weeks ago. -Matt

Hi * users, I want to know if STUN suport is available with Asterisk. Kindly let me know. I have posted this also in DEV list but none replied to me. thanks, Somesh -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050808/e26855c9/attachment.htm

The cheap ones on EBay won't work with the SC420 server. I have one and can't make any of the clones work. I do have one TDM40B card for analog stations that works well. The problem with the SC420 is that it won't let you set the interrupts yourself and you end up with interrupts being shared. =============================================================== Message: 26 Date:

Hello, I am trying to let someone know that is being called from a specified location. For that, the command: exten => _107.,1,Dial(SIP/SIPGW/${EXTEN:3},30,A(Anounce.gsm)) should let the called person hear Anounce.gsm as soon as he/she answers. (Only calls with prefix 107 are given this notice). The call proceeds fine, but no one hears AnounceSPF.gsm. I tried putting this file in every

I just bough a Sipura SPA-2100 to use with Asterisk. When I use the analog handset plugged into the SPA-2100, the person on the other end can hardly hear me. I check the SPA-2100 setup and their is no mic/spk gain control. Is this a problem with the SPA-2100 or with Asterisk? Any way for asterisk to compensate for the poor audio level (if the problem is the SPA-2100)? Thanks, Mike

I've just gotten off a skype conference call and it pisses me off that the quality of skype is higher than my asterisk calls. Is there such a thing as a super high bandwidth codec? In a situation that you have the bandwidth to share is there something that I can use for important calls when the situation warrants it? TIA, Dean -------------- next part -------------- An

Tim, I reported a protocol data stream error on 9/8/2002 Deatils are in my postes at that time rsync still fails everytime I run it. This time I added three v's as you just suggested and rsync just hung with two tasks in execution thus: root 3178 0.1 0.9 1796 868 pts/1 S 09:57 0:00 /usr/bin/rsync -vvva --delete --exclude=root/snapsh root 3179 0.0 0.6 1628 632 pts/1

is it possible to register asterisk in a sip proxy as if it were a terminal (like a cisco ATA)? how? Thanx Jenna ;) ___________________________________________________________ 1GB gratis, Antivirus y Antispam Correo Yahoo!, el mejor correo web del mundo http://correo.yahoo.com.ar

I'm trying to get this zap dial to work. I want to send DNIS and ANI to other system (ZAP/g2) at answer, while the caller hears ring (RBT). I hear the RBT, but callerid and exten is not sent to other T1 - The ZAP/g2 T1 is standard D4, SF, E&M Wink Start. - At ZAP/g2 wink, asterisk should send DTMF "*ANI*DNIS*" exten => _XXXX,1,NoOp,${CALLERID} exten =>

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How come an outgoing call using my TDM400P immediately say the call is answered? I'd like to be able to detect when the call is actually picked up, is this possible? If this is normal with analog cards, does the same thing happen with T1 cards? -L ____________________________________________________ Yahoo! Sports Rekindle the Rivalries. Sign up for Fantasy Football

hello i want to use SIPGetHeader application in asterisk-1.0.9. Jul 2 00:04:33 WARNING[19575]: pbx.c:1293 pbx_extension_helper: No application 'SIPGetHeader' for extension (default, 2000, 1) Any one using this __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

Hello. How can I check if the RTP traffic between two channels is bypassed? Some * console command? Thanks.

Hello All, I have an iaxy(new version), and while it does the job well, there is one thing I am looking for. I want to be able to specify a dns name on the config, not an ip. This does not seem to work if I try to set it as such. Has anyone come up with a workaround or solution to this? I want to be able to put it on the net, or travel with it, plug and go. The issue is that I am using a

We've just setup our A@H server, with our quad port card. Everything works well so far. One thing I notice is that when I leave the handset on the hook and dial a #, all is well. If I pick up the phone and dial, it cuts off at 10 digits, which is a problem if I need to dial 1+area+phone # (12 digits). The phones are Poly Soundpoint IP 600's. I'm wondering if I've missed a

via google, I found the reference regarding digit maps for the Polycom phones: http://lists.digium.com/pipermail/asterisk-users/2005-January/082884.html But I don't see any instruction for prepending digits to the number dialed. Does anyone know how to prepend a digit to the number dialed (from the Polycom side, not Asterisk)? I can do this pretty easily on a Sipura. i.e. Say I want to

Hi, Please help me through to overcome this error. I have my asterisk server and a xlite client in 172.16. network. I opened up port 5060 in my enterasys wireless router for my asterisk server and also opened port 8000 and 8001 for my xlite client. When i tried calling a remote system which successfully connected to my asterisk server i'm getting the error as " call failed: 499 Not

Hi all, When I was making calls from an IP phone, through a X100P, to PSTN, the following error was encountered. -- Executing Dial("SIP/25086937-aa6c", "Zap/1/91713545") in new stack -- Called 1/91713545 -- Zap/1-1 answered SIP/25086937-aa6c Aug 6 00:12:53 WARNING[3983]: chan_zap.c:4717 zt_indicate: Don't know how to set condition 16 on channel Zap/1-1 --

Does anyone know of a way to make a standard analog phone plugged into an FXS port do something other than get a dialtone when you pick it up? For example, if the phone should automatically ring someone or play a greeting when picked up without having to enter an extension? - Robert -------------- next part -------------- An HTML attachment was scrubbed... URL: