similar to: SIP vs. SIP :-(

I can dial from an asterisk host to another one via FreeWorldDialup, on the other side DISA service answer to me and i can ear dialtone. But i cannot send DTMF and dial an extension on the DISA enabled asterisk.....i've tried rfc2833 and inband...but nothing....any tips ??? Thanks, -- Igor Barsanti GPG Public key available at http://pgp.mit.edu

Thanks for taking a look greg and hank. This seems to be getting bettre everyday..help please My sjphone is running on the same box as asterisk...i believe then the red hat firewall should not be a problem. Whenever i dial from CLI i get ######### Executing Goto("OSS/dsp", "default|s|1") in new stack -- Goto (default,s,1) -- Executing Wait("OSS/dsp",

Hi, I have a problem with IAX accounts... I set up a few months ago an Asterisk server, with mysql support to load iax accounts. Settings seems fine because apparently the system works as expected. Yesterday I tried to add another iax account in the iax.conf directly. And I have a problem with this new account (named 444). I can authenticate from 444 to the server, and I can receive calls from

Hi :) I've had all this working before, but I'm revisiting it, and in short, I currently have huge problems receiving incoming calls. I've been trying with both FWD and voiptalk.org. I'm running CVS HEAD of asterisk, zaptel and libpri as of yesterday afternoon. Would someone mind helping? :) My machine is 10.0.0.1 on my LAN, but the ADSL router has 10.0.0.1 set as the 'DMZ

Hi Everyone, I've got a Cisco 3810 rig with 6 analog FXS ports, and ethernet. I'm wondering in anyone has got one of these suckers to work with asterisk in such a way that each FXS port has it's own extension. It speaks SIP, and I can send calls from asterisk out to it, but can't figure out how to get it to pass username & pw to asterisk when I try to configure it as a

Hi, Every time I dial into my asterisk box i hear nothing but asterisk jittering. The following is an example of what I get on the asterisk CLI Thanks *CLI> DEBUG[81926]: File chan_sip.c, Line 3826 (check_user): Setting NAT on RTP to 0 DEBUG[81926]: File chan_sip.c, Line 4807 (handle_request): Check for res DEBUG[81926]: File chan_sip.c, Line 952 (find_user): Call from user

Hi there, do I have a local problem, or is registration at IAXTEL impossible at the moment? "iax2 show registry" permanently shows a TIMEOUT for 69.73.19.178. Philipp

Hi all, I have the following setup: UAs ------------SER ------------------------ ASTERISK ---------------------GNUGK --------------- GWs SER is configured to route call requests from UAs to Asterisk. Asterisk is configured to receive the call on SIP channel and dial out to GNUGK over H323 channel. The problem I'm facing is that asterisk sends out the call request to GNUGK and times out

I was able to get app_prepaid working, but unfortunately I am getting one way audio on the phone that I was placing the call from. It is behind NAT. It appears that the app_prepaid is not taking this into consideration since I see: Jun 18 17:46:25 DEBUG[1133742896]: chan_sip.c:4130 build_route: build_route: Contact hop: <sip:7708183799@192.168.1.101:5060;line=jet7pbic> Jun 18 17:46:25

The IAX2 channel is to IAXmodem. The SIP extension is an ATA with a fax attached. Nov 28 15:30:20 DEBUG[2997] chan_sip.c: build_route: Contact hop: Nov 28 15:30:20 VERBOSE[3276] logger.c: -- SIP/2201-090995f0 answered IAX2/24729-2 Nov 28 15:30:20 DEBUG[2995] chan_iax2.c: Ooh, voice format changed to 8 So what does this mean? The fax works just fine. I am just trying to tune up my dialplan.

Hi i have an issue getting agentLogin working /etc/asterisk/queues.conf member => Agent/1001 member => Agent/1002 extension.conf exten => 110,1,Wait,1 exten => 110,2,AgentLogin() now, i call 110 by a firefly client, trying to login in as 1001 agent: Aug 12 16:31:36 DEBUG[1103408048]: chan_sip.c:4423 build_route: build_route: Contact hop: <sip:sip3@192.168.1.151:5060> --

Hello folks, I'm having trouble configuring Asterisk to play an "invalid extension" message to anyone dialing an undefined extension. First I tried using the 'i' pseudo-extension, but it didn't work at all; searching the wiki I found that page: http://www.voip-info.org/tiki-index.php?page=Asterisk%20i%20extension where it basically says that the 'i'

Hello All I am trying to answer incoming SIP calls, first, by dialing an extension, thence into voicemail, which works; and secondly by going straight into voice mail which does not. The extension.conf that works is like this; [incomingSIP] exten=>_.,1,Dial,Zap/2|1 exten=>_.,2,Voicemail,u5152 exten=>_.,3,Hangup the extension.conf which does not is like this; [incomingSIP]

I can't do threeway from my Grandstream phone. Looking through the server config files, I figured out why - zapata.conf has Threeway turned off for the channels I use. I do my work on someone else's Asterisk box and don't want to modify zapata.conf for several reasons, the biggest being that the guy who owns the box has a couple clients using it and I am deathly afraid of breaking

How can I make it more readable? Name/username 601/601 123456789/123456789 voipbuster/abcd 601 = hotline 123456789 = Peter Pan only voipbuster/abcd is easy read/understandable! bye Ronald Wiplinger

Can anybody please answer this. Both clients are behind different NAT's. One of them starts a SIP call to the other through Asterisk. Asterisk sets up the call. Issues reinvite and connects them together. After this point does the media stream flow through Asterisk or Peer to Peer? Does such a call use any system resources of Asterisk server after connection? Thank you in advance.

I'm trying to get asterisk working w/ packet8 (incoming and outgoing) and a dta310 so I can have more control over voicemail. I've looked at the data stream coming from the dta310 and from packet8, but I haven't managed to get the phone to actually place invites through asterisk. On the asterisk end with chan_oss.so, I can make it dial and I hear ringing and the first second of

Hello. Sorry to repost this message but, I don't have the original message in my inbox nor in my sent box. Well, last week I posted a problem I am having trying to use an asterisk server use a voip provider and a pstn. Pstn works fine but, I cant even connect to my provider's server. I don't know what I'm doing wrong. I tried using a soft phone and I'm able to register and

We are using the following to record conversations. exten => _1XXX.,1,SetVar(CALLFILENAME=call_to_${EXTEN:1}_dated_${TIMESTAMP}) exten => _1XXX.,2,Monitor(wav,${CALLFILENAME},m) exten => _1XXX.,3,Dial(IAX2/4506:zj5S3A5a@nl.voipgate.nl/${EXTEN:1}) exten => _1XXX.,4,Congestion exten => _1XXX.,104,Congestion This was working previously to record both sides of the conversation but now

I have searched all over the web and have not really found anything related to this error.... The only thing I found is related to a system stops responding on DTMF, which does not happen here... THe following is the output from the CLI: *CLI> 2004-08-09 17:36:29 DEBUG[229390]: chan_sip.c:2332 sip_alloc: Allocating new SIP call for 640E2D47-E98B11D8-8FDBE54A-5FB5A0CB@65.67.76.30