similar to: Regular crashes

Thanks Paradise, this seems to have worked a treat!!! I commented out the: exten => 110,hint,SIP/110 lines which were in extensions_additional.conf for each sip extension I had. This seems to have stopped the crashes which were previously 3-5 times a day, now: System uptime: 1 day, 18 hours, 10 minutes, 3 seconds Interestingly it had the knock on effect of fixing another problem I had

What does this mean? Asterisk 1.2.13 talking to 1.4.0. (response from 1.4.0.) Yuan Liu

Hello there! I'm working on some modifications on Asterisk to adapt it to our needs considering some particular demandings of the infraestructure we want to provide. Two of these modifications are: 1- A proprietary configuration driver that will communicate with a server that will be the source of information for the entire infraestructure; and, 2- A call control application that will be

Hi list, My need is to append a site specific parameter to the Contact: header on all INVITEs exiting * via a SIP trunk. I'd like it to look something like this: Contact: <bob:3125551212 at 10.10.10.10;SITE-ID=us.here> where SITE-ID=us.here is set in a config file that * parses on startup. Or in a Dial() command option? Or I don't care exactly how. :-) It is possible to

Hello, How can I get peerip, recvip, from, uri, useragent, peername, t38passthrough variables in (within) my custom Asterisk application? I can't use chan_sip.c internal structures (such as sip_pvt) in my custom application, because there's no chan_sip.h and I can't include it into my application (maybe there's other way?). I can do like this: exten =>

I know I'm getting tired of looking at code, but why is the function register_verify defined in 2 different files? chan_iax2.c line 3860 static int register_verify(int callno, struct sockaddr_in *sin, struct iax_ies *ies) chan_sip.c line 4869 /*--- register_verify: Verify registration of user */ static int register_verify(struct sip_pvt *p, struct sockaddr_in *sin, struct sip_request *req,

Hi, I'm using Asterisk (1.2.4) as a voicemail system for our Softswitch. When forwarding a call to Voicemail, here is somehow what the softswitch sends to Asterisk : In INVITE : Vm Phone Number ( to route the call ) In To : Person who has been called ! In From : Person who was calling ! Of course, I need to send the call into the "Called User" Mailbox (Thus To SIP header) ! So

I did some searching in the archive, but found only one message with this same question and no answer. Hopefully it's a simple config problem. When the Caller-ID is delivered, it is surrounded by double-quotes, like this: "ATA-57 1" On long caller-id strings, the last character is cut off to make room for the leading double-quote: "BudgeTone 1234 instead of BudgeTone

Dec 1 12:08:43 WARNING[6189]: channel.c:495 ast_channel_walk_locked: Avoided deadlock for 'SIP/2502-6303', 10 retries! Dec 1 12:08:44 WARNING[6189]: channel.c:495 ast_channel_walk_locked: Avoided deadlock for 'SIP/2502-6303', 10 retries! Dec 1 12:08:44 WARNING[6189]: channel.c:495 ast_channel_walk_locked: Avoided deadlock for 'SIP/2502-6303', 10 retries! what does this

Can someone help me with upgrading to the lastest version. I am using the same sip.conf file, but the headers have changed and registration fails. Has something change in the conf file that would cause this? Notice 1.2.5 has no Authoization at all... Regards, Jason Version 1.0.9 --------------------------- REGISTER sip:voip-ca35323.ocn.ne.jp SIP/2.0 Via: SIP/2.0/UDP

Hi, I'm testing Asterisk with a new provider. On calls to US toll-free numbers, there is no audio (calls to normal numbers are ok). In response to a valid INVITE from Asterisk, something like this is received: SIP/2.0 183 Session Progress v:SIP/2.0/UDP [my public IP]:5060;branch=z9hG4bK62d91cea CSeq:103 INVITE i:7a1791cf52d6f3dc2d12b208051d0a21@[provider].com f:"Test User"

----- Original Message ----- From: <asterisk-users-request@lists.digium.com> To: <asterisk-users@lists.digium.com> Sent: Wednesday, December 01, 2004 7:07 AM Subject: Asterisk-Users Digest, Vol 5, Issue 6 > Send Asterisk-Users mailing list submissions to > asterisk-users@lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, visit >