similar to: Unblocking incoming SIP

I have been trying to locate the patch that is supposed to cure the problem of hearing sound from the previous call when dialing through i4l and an hfc card. Does anyone have it? It is mentioned briefly in this post: http://lists.digium.com/pipermail/asterisk-users/2003-February/007530.html Thor

I have set up asterisk with an ISDN card using i4l. When I place a call from ISDN to a SIP client, there is about a one-second delay from a word is spoken to it is heard at the other end. The funny thing, is that the last second or so of each call is saved somewhere in the depths of Asterisk and then played back at the beginning of the next call. I have read it is a known problem, can

Hello list, I have been running Asterisk CVS for a good while. When I try to install 1.0.3, asterisk won't start. Below are the last few lines of output before Asterisk crashes. I ran "make samples" to start with a fresh setup. [app_read.so] => (Read Variable Application) == Registered application 'Read' [app_alarmreceiver.so] => (Alarm Receiver for Asterisk) ==

I run an Apple OS X workstation and I've got a server on the same LAN that's both a webserver and an Asterisk PBX. I wanted to be able to originate calls in the OS X Address Book application, and have Asterisk dial them and connect them to the phone on my desk. I've assembled a system that uses AppleScript to connect, via XML-RPC, to a web application that, in turn, connects to

Hello folks, i upgraded to the actual CVS head from yesterday (27.5.) but can not get incoming SIP calls from my provider (sipgate). If someone calls my number, my asterisk responds with the following error: May 27 21:30:21 NOTICE[1114606512]: chan_sip.c:6351 handle_request: Failed to authenticate user "<CallerID>" <sip:<CallerID>@217.10.66.11>;tag=as38e9693c I

I am trying to compile chan_capi 3.3.4a, but I end up with lots of gibberish. Near the top it states that capi20.h doesn't exist. Searching for the file, several show up: # find / -name capi20.h -print /usr/src/linux-2.4.21-144/include/config/isdn/capi/capi20.h /usr/src/linux-2.4.21-231-include/smp/include/config/isdn/capi/capi20.h

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I'm wondering if anyone else has gotten something similer to this? I had FWD working fine on the asterisk box, then all of a sudden it just stopped working. I get the following errors (just keeps looping) *CLI> DEBUG[1125329600]: File chan_sip.c, Line 527 (__sip_ack): Stopping retransmission on '6dc8436c7c568eea75fffdc75478ed54@142.55.31.179' of Request 102: Found

Sorry this is a bit of a newbie question, I've been at this for a few months and still have not quite figured this one out. I've been able to setup one itsp (incoming calls) (sip account) with a register line like this: register => nnnnnnn:ppppp@sip.provider.net -or- register => nnnnnnn:ppppp@sip.provider.net/nnn to come directly into an extension in the dialplan It seems that

Sirs, Thank you for your quick response. But when i try to make a call to FWD the following error appears: For example, when i call to 612 (a service number of FWD) -- Executing Dial("SIP/Phone4-e85b", "SIP/612@fwd.pulver.com|90|Ttr") in new stack -- Called 612@fwd.pulver.com -- Got SIP response 500 "I'm terribly sorry, server error occured (1/SL)"

Hi forum, I have been fighting days and days configuring FWD and asterisk with NO success I have the following scenario. My sister in Spain with FWD dialup client My question is if she can dial my FWD dialup number, which is registered in Asterisk and the call being forwarded to ring my IP Phone. Spain LAN FWD

Hi, Is it possible to use asterisk with Free World Dialup (FWD) ? Did someone manage to make it work? how? Best, -P -- __________________________________________________________ Sign-up for your own personalized E-mail at Mail.com http://www.mail.com/?sr=signup CareerBuilder.com has over 400,000 jobs. Be smarter about your job search http://corp.mail.com/careers

I have a small HUGE problem with *. I have installed * but I have 2 problems. 1 - Making call to FWD. 2 - Receiving call from FWD More info of the problem at the end. Here is the sip.conf file. ; ; SIP Configuration for Asterisk ; [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = sip ;default Default for incoming calls register =>

Can multiple FWD accounts be registered? I have the following output in my sip.conf file: register=74928:xxx@fwd.pulver.com/74928 register=75160:xxx@fwd.pulver.com/75160 register=74573:xxx@fwd.pulver.com/74573 [fwd-74928] type=friend secret=xxx username=74928 host=fwd.pulver.com [fwd-75160] type=friend secret=xxx username=75160 host=fwd.pulver.com [fwd-74573] type=friend secret=xxx

Hi, after reading about it on the list I decided to set up a Free World Dialup account. For those of you who don't know, that is a sip proxy where you and your friends can singn up free and then you can just connect to it with any sip client and call anybody that is registered for free. Pretty much like iaxtel (I belive that was the name of it) for the iax protocol. It even supports clients

I can't get my Asterisk to register/place calls with FWD. Here's what I have in my SIP.CONF: register => 11111@fwd.pulver.com/11111 [fwd] type=friend secret=somesecret host=fwd.pulver.com username=11111 fromuser=11111 fromdomain=fwd.pulver.com I'm using CVS version of Asterisk, checked it out last week. I get authenticate error when registering with fwd, and all my calls to

Using DIAX softphone which seems to be working OK can get to VM/echotest etc in the demo context Am trying to setup FWD but get the following problems Can hear it ringing when dialing FWD no 612 for time. Connects but no sound from remote end. Does anyone have any suggestions. Softphone on 192.168.0.2 asterisk on 192.168.0.3 Netgear RP114 doing NAT to the internet port 5060 being forwarded to

Hey all, I'm trying to get my Asterisk server up and running on fwd.pulver.com just to get the hang of it until I get my FXO card in a couple of days. It seems to connect but that's about it. If I try to dial into it from another fwd # it says user is not online. In sip.conf I have the following added: register => xxxxxx:xxxxxx@fwd.pulver.com/489125 [fwd.pulver.com] type=friend

Still getting the same error. Apr 29 11:57:49 WARNING[1125329600]: chan_sip.c:503 retrans_pkt: Maximum retries exceeded on call 6b8b4567327b23c6643c986966334873@211.28.255.135 for seqno 102 (Critical Request) please advise anyone!!!!!someone!!! jai

FWD (Free World Dialup) allows any SIP call to US toll free numbers via * 18xxzzzyyyy at fwd.pulver.com This works WITHOUT the need to be registered at FWD so in my dialplan I have something like: exten => _8.,1,Dial(SIP/fwd.pulver.com/*${EXTEN:1},60,r) exten => _8.,2,Hangup And I just dial 8-1-8xxyyyzzzz and presto ... calls go through just fine 99% of the time. I'm wondering if