similar to: Early media after a dial command

I've got a friend who's spending 6 months on the other side of the world. So before he left I configured him a softphone on his laptop to connect to my asterisk so he can call home free of charge. Unfortunately, he just found out he has horrible internet connection. Bandwith and latency is ok, the problem is the stop almost all connections. He has to connect to a proxy server for his web

Hi! I have an T1 line for using with IVR AGI. I receive the calls in my T1 but my dialplan has an error but my extensions doesnt have the error that show me asterisk. I dont know from where asterisk take extension 8 and how is playing ss-noservice because in my dialplan is not exist. Any help or any cluees? Verbosity was 5 and is now 7 -- Starting simple switch on 'Zap/1-1' ==

Hi all, I have a connect between a siemens hipath & Asterisk system over PRI The connection works perfectly I can call from the Hipath to an Asterisk Extension. I want to allow the hipath extensions to dial out over a SIP trunk on asterisk but I keep getting "The number you have dialed is not in service" In this CLI I'm dialing from hipath extension 9 (prefix for sip trunk)

> Post your extensions.conf and what's on the CLI (asterisk -r) As requested: # cat /etc/asterisk/extensions.conf [incoming] exten => s,1,Answer() exten => s,n,NoOp(CallerID is ${CALLERID}) exten => s,n,NoOp(DID is ${DNID}) exten => s,n,Background(enter-ext-of-person) exten => 1625,1,Playback(digits/1) exten => 1625,n,Goto(digits/1) exten => i,1,NoOp(CallerID is

Hi, I can't receive/send calls with Asterisk. Could someone please give me a few pointers on my configuration? Regards, Norman Zhang ; sip.conf [general] disallow=all allow=ulaw port=5060 bindaddr=0.0.0.0 externip=x.x.x.x localnet=192.168.22.0 mask=255.255.255.0 context=inbound-sip maxexpirey=180 defaultexpirey=160 tos=reliability srvlookup=yes register =>

Hello to all, Is there an "how-to" for asterisk and setting up a t38 fax gateway (SIP) ? I look at http://bugs.digium.com/view.php?id=5090 to patch asterisk chan_sip.c file. What are the next steps to get a t38 fax gateway with asterisk ? Regards Harry PS: I use hylafax server. ___________________________________________________________________________ Faites de

Hello, I read http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+SIPredirect I wish to configure asterisk as a redirect server. I have badly understood this command . ASTERISK | sip agents nated ======SER When sip agents send INVITE to the proxy i want ser send this one to ASTERISK . How asterisk can send 302 messages with the good info in

Hello, I patched asterisk patched with the latest t38 support . I would need some people for tests. Regards harry ___________________________________________________________________________ Faites de Yahoo! votre page d'accueil sur le web pour retrouver directement vos services pr?f?r?s : v?rifiez vos nouveaux mails, lancez vos recherches et suivez l'actualit? en temps r?el.

Hello, I 've installed both cacti and res_snmp for monitoring. Does res_snmp is able to send snmp traps when hardware is out of service or others status ? Harry ___________________________________________________________________________ Faites de Yahoo! votre page d'accueil sur le web pour retrouver directement vos services pr?f?r?s : v?rifiez vos nouveaux mails, lancez vos

I've noticed I have an issue with my Dialplan ... apparently instead of a busy signal when the caller is busy it falls through and gets a Congestion... What's the proper syntax for this, reorder tone when there is a reorder and busy when there is a busy... SBC is a T1/PRI. [macro-sbc-outdial] exten => s,1,Dial(${ARG1}/${ARG2}) exten => s,2,Congestion exten =>

Hello, What's wrong ? make install ..... options torisa base=0xd0000 alias char-major-196 torisa alias wcfxs wctdm alias wct2xxp wct4xxp if [ -d /etc/modutils ]; then \ /sbin/update-modules ; \ fi depmod: *** Unresolved symbols in /lib/modules/2.4.27-2-386/misc/wctdm24xxp.o depmod: *** Unresolved symbols in /lib/modules/2.4.27-2-386/misc/zaptel.o depmod: *** Unresolved symbols in

I am trying to make an in/out trunk group comprised of 4 DS0's using E&M Wink signalling. The first four channels of a DS1 on a T100P are being used for the group. Outbound calls work fine, but inbound calls fail. The other 20 DS0 channels are used for a PRI. Does the configuration shown below look okay? I've tried setting 'immediate => yes' without success, but it

I am setting up a box with a T100P. Everything is going well. The company I am working with has their one phone switch gear. They provisioned me a E&M Wink T1. Cannot do PRI unfortunately. We chose E&M so we could pass an unlimited number of DIDs to the trunk as apposed to FXS loopstart signaling. I can make outbound calls no problem, but I am having problems with the dial plan for inbound

I'm trying to use OpenBTS with Asterisk. I have two phones that are connecting to OpenBTS correctly, but on the Asterisk side the phones can't call each other. I followed this guide: http://gnuradio.org/redmine/projects/gnuradio/wiki/OpenBTSSettingUpAsterisk I set up two phones in sip.conf and extensions.conf. In my SIP output I see this: WARNING[1689]: app_dial.c:2041 dial_exec_full:

Hi, Can I make e&m wink start lines just wait for digits - instead of going to default? Someone else cleared a similar problem (as described below) on an fxo port with "usecallerid => no" but it is not doing the trick for me. In this case the line when straight to default which would be ok also. John I posted the stuff below about a week ago... I set up a t1 from my sys75 to

I do part time consulting work. I need to setup an asterisk system to allow me to record both inbound and outbound calls to clients. I have one client that is just a PITA. The client has changed their mind three times so far and we are at step one. I have a spare slackware box and a seperate phone line for the consulting work. I have MCI Neighorhood as my carrier. What I need to know is: 1.

Hello, I have an error when installing AMP, when I do ./install_amp --debug, it show me : Connecting to database..FAILED [DEBUG] [nativecode=Can't connect to local MySQL server through socket '/var/lib/mysql/mysql.sock' (111)] ** mysql://user:pass@localhost/asteriskamp Try running ./install_amp --username=user --password=pass (using your own user and pass) [FATAL] Cannot connect

hello, Is there somebody wit a h323 terminal ? ___________________________________________________________________________ Faites de Yahoo! votre page d'accueil sur le web pour retrouver directement vos services pr?f?r?s : v?rifiez vos nouveaux mails, lancez vos recherches et suivez l'actualit? en temps r?el. Rendez-vous sur http://fr.yahoo.com/set

Hi does asterisk act as SIP proxy ?, like SER any documents if does, will be great help ram -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060516/9fce4aca/attachment.htm

Hello every body. I have this PCI card : DM/V1200-4E1 spec in this site: http://www.intel.com/network/csp/products/3967web.htm Can i use it with Asterisk, is it compatible ? Thank you in advance. ___________________________________________________________________________ Faites de Yahoo! votre page d'accueil sur le web pour retrouver directement vos services pr?f?r?s : v?rifiez