similar to: Asterisk & SIP

From: "Nik Martin" <nmartin@radiancetech.com>> To: <asterisk-users@lists.digium.com>> Subject: RE: [Asterisk-Users] asterisk compile problem Date: Wed, 14 Jul 2004 09:22:38 -0500 Organization: Radiance Technologies, Inc. Reply-To: asterisk-users@lists.digium.com Fletcher Bonds wrote: >> Hello all >> >> As of 5pm PST today (7/13), I pulled

I'm getting the following errors out of my asterisk console: Jul 14 15:12:12 WARNING[40966]: chan_sip.c:673 retrans_pkt: Maximum retries exceeded on call 5d33ce4422e629921875e8a7609c1603@0.0.0.0 for seqno 141 (Critical Request) Jul 14 15:12:26 NOTICE[40966]: chan_sip.c:3902 sip_reg_timeout: Registration for '451411@fwd.pulver.com' timed out, trying again Jul 14 15:12:26

I am wondering how the g729 licenses are done during calls. If I have N licenses for g729, and N are in use and an additional call comes in that requests N+1 to be in use, how does asterisk handle that call? Does it dump it? Does it negotiate another codec automagically? Basically what happens to that call, obviously it wont (shouldnt) let you use more licenses than you have available, but

I'm trying to test IAX2 with FWD It registers fine but when I try to receive the call I get: chan_iax2.c:476 iax_error_output: Ignoring unknown information element 'Unknown IE' (38) of length 1 Jan 25 18:02:12 WARNING[114696]: chan_iax2.c:476 iax_error_output: Ignoring unknown information element 'Unknown IE' (39) of length 1 Jan 25 18:02:12 WARNING[114696]: chan_iax2.c:476

Dear all, I have an asterisk sip issue which I don't believe is unique. I use a registrar (sipgate.co.uk) where I have 3 different accounts. These accounts provide me with three seperate local phone numbers which allow me to allocate them to seperate users. By using just one of these accounts I can set asterisk up to send and receive calls no problem. However, when I start to introduce an

Could someone please point me to the proper url to register for a fwd acount and get a fwd number. I couldn't find it at www.freeworlddialup.com or fwd.pulver.com Thanks, -Galt -- "They that would give up essential liberty for temporary safety deserve neither liberty nor safety." Benjamin Franklin

Initially I'm setting up asterisk to do SIP only, no cards, no connection to PSTN or ISDN - just receiving 'calls' via SIP, answering them and playing a message.. Can anyone point me to a walkthrough that is focused on SIP alone? Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Sipdiscount has replaced their asterisk servers for another thing. Then, no more iax. Ok, but I can't make calls using sip also... I'm getting a "forbidden" error when using sip1.sipdiscount.com. Anybody got it working? -- Alejandro Vargas

When I try to call out to FWD over IAX2 I get: Call rejected by 65.39.205.121: Unable to negotiate codec I'm using asterisk-1.0.5 (the same settings works fine with *0.9) I've standard settings in iax.conf [general] bindport=4569 register => xxxxx:xxxxxx@iax2.fwdnet.net [iaxfwd] type=user context=fromiaxfwd auth=rsa inkeys=freeworlddialup disallow=all allow=ulaw -- #Joseph

Hello, Sorry for what may be a basic question, but I have spent a number of hours trawling the forums without resolving the problem, and hence this post. I have just started to dabble with Asterisk, as much for the learning than anything else. I created an account on FWD and used the Asterisk settings that the FWD web site recommends at

I am looking for a voip provider that provides good rates (below 5 cents/min or unlimited) to UK NCFA numbers. Braodvoice advertises they do unlimited to NCFA but does not have the ability to actually termiate those calls as per the CTO Nathan Stratton, and last he said they dont even have contracts in place to get service provisioned for that. As such I am looking for another provider to take

Hi all, I am new to this list... Can I use Asterisk as a SIP Proxy and as a SIP to H.323 translator? I want to implement PC-to-Phone calls in the following topology (for signalling): SIP Softphone --> Asterisk --> Gatekeeper H.323 ---> Gateway H.323 ---> PSTN The RTP audio packets would go direct through Softphone to gateway. Does someone have a configuration file example of

Hi Everyone: I have a THREE questions. What is a sip proxy and what is the benefit of having one with Asterisk? I am well aware that we have a sip channel in Asterisk and that we have SIP registration. I am not sure why you would need a SIP server and OR a registration server. Second question, with Asterisk are you able to do video on VOIP video phones? Last question, does

Has anyone successfully used a standard bluetooth enabled system to connect to a standard bluetooth enabled mobile phone (not the bluetooth to FXS converters) to create an audio path for phone calls with asterisk, if so is there a writeup on what was done so that others can replicate this. What I am thinking is that via alsa/oss/whatever you should be able to use the bluetooth audio channel as a

Recently I have been building LLVM and Clang from the distribution using gcc 4.9.2. With the new 5.0.0 rc2 that is giving warning messages during the compilation. I have been trying out some other compilers. gcc 5.2 with -std=c++11 This works, although there are still some warnings. gcc 6.4 and gcc 7.1 fail with errors such as this in building libcxxabi as follows: Command:

I am trying to make 1 soft SIP UA behind NAT connect to a public hard CISCO UA via a public asterisk server. The CISCO UA can hear the voice from the SIP UA but not vice versa. I do set nat to yes for the soft phone. Any help would be greatly appreciated. Below is my sip.conf [general] port = 8060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0 ; Address to bind to (all

Hola, I am trying to isolate the memory leak I suspect in a mailman installation ? I found: http://blogs.sun.com/sanjeevb/date/200506 It gives an error: god at irt-smtp-02:~ 9:21am 65 # ./memleak.d 10312 dtrace: failed to compile script ./memleak.d: line 3: probe description pid10312:libc.so.1:malloc:entry does not match any probes I am on SunOS 5.10 Generic_127112-07 i86pc i386 i86pc Are

I'v a sip client and a sip trunk to FWD: [general] port=5060 context=default tos=reliability disallow=all allow=ulaw careinvite=no [freeworlddialup] context=default type=friend username=MYUSERNAME secret=MYPASSWORD host=fwd.pulver.com [igor] type=friend callerid="Me" host=dynamic dtmfmode=rfc2833 careinvite=no When i try to call a FWD number from SIP client i obtain a lot of

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

Hello !! I saw in FWD site a phone on the web.. (click 612 link) http://www.freeworlddialup.com/advanced/beta_programs I?d like to have this application in my intranet.. click on my name, than calls my number.. I?d also like to see that phone on the web... as an option How can I do that ? Is it possible to download ? Any related link ? Thanks Andrei.