[asterisk-users] netsock error? some sip clients crashing!

Hello I have a weird behaviour with our local GSM (3G) provider -- several SIP clients crash on the android phone, when switching to 3G network, and in asterisks logs it looks like this - client registers on server successfull and then crashesh immediately. Here's suspicious part of asterisk

[asterisk-users] 1.8.5 CLI colors are gone?

Hello My CLI of 1.8.5 is black and white? How do I re-enable the color highlighting? Thanks Nick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory

[asterisk-users] Name or service not known

Hi, my log is full of errors from this mobile user: -- Registered SIP '0010106' at 212.93.97.135:7759 [2011-05-17 17:44:05] NOTICE[21456]: chan_sip.c:19804 handle_response_peerpoke: Peer '0010106' is now Reachable. (381ms / 1ms) [2011-05-17 17:44:06] ERROR[21456]: netsock2.c:245

[asterisk-users] 1.8.4 keeps quitting console by itself

Hi! I've noticed 1.8.4 keeps quitting console by itself. Is this a bug or feature? :) Nick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar

[asterisk-users] 1.8.4 quitting console

actually i just noticed that it quits console because asterisk restarts itself after: [2011-05-16 13:48:45] ERROR[11106] tcptls.c: Unable to connect SIP socket to 192.168.1.108:5060: Connection timed out -- _ -- Bandwidth and

[asterisk-users] log full of Name or service not known

Hi! Here's a user with mobile phone - however why does it treat this as ERROR ? I have a log full of that --- -- Registered SIP '0010106' at 212.93.100.181:3698 [2011-05-12 16:07:57] NOTICE[30258]: chan_sip.c:19679 handle_response_peerpoke: Peer '0010106' is now Reachable. (212ms /

[asterisk-users] special control 16

is 0x8 (alaw) read/write = 0x8 (alaw)/0x8 (alaw) [2011-03-28 18:22:27] WARNING[22502]: chan_sip.c:6064 sip_write: Asked to transmit frame type ulaw, while native formats is 0x8 (alaw) read/write = 0x8 (alaw)/0x8 (alaw) -- Nick Ustinov

[asterisk-users] blind transfer from AGI triggered call - dropped

Hi! Maybe someone could help me out? When a call is routed via a2billing AGI and user does a transfer, the call is dropped. If the trunk is called directly everyhing works. Here's a direct scenario (working fine): [pbx01] exten = 101,1,Set(__TRANSFER_CONTEXT=pbx01) exten =

Re: [asterisk-users] chan_sip.c: Failed to parse contact info

Well, it has disappeared in further builds ;) Thanks 2011/3/16 Leif Neland le...@neland.dk: Den 19-01-2011 00:19, Nick Ustinov skrev: Hello! I have just upgraded to asterisk 1.8.2.1 and see some weird messages in log when client tries to register: [2011-01-19 00:52:47] WARNING[25624

Re: [asterisk-users] Asked to transmit frame type slin, while native formats is 0x8 (alaw) read/write = 0x4 (ulaw)/0x8 (alaw)

These are the same for sip users and trunks disallow=all allow=ulaw allow=alaw allow=gsm allow=g729 Who is asking to transmit frame type slin ? Nick On Thu, Mar 10, 2011 at 1:02 AM, Paul Belanger pabelan...@digium.com wrote: On 11-03-09 02:26 PM, Nick Ustinov wrote: Using asterisk 1.8.4-rc2

[asterisk-users] Asked to transmit frame type slin, while native formats is 0x8 (alaw) read/write = 0x4 (ulaw)/0x8 (alaw)

Hello! Client is using ulaw, however server sometimes fills the log with following: [2011-03-09 21:23:07] WARNING[27204] chan_sip.c: Asked to transmit frame type slin, while native formats is 0x8 (alaw) read/write = 0x4 (ulaw)/0x8 (alaw) [2011-03-09 21:23:07] WARNING[27204] chan_sip.c: Asked to

[asterisk-users] Ignoring RTP 2833 Event: 0000009f. Not a DTMF Digit.

Hello ! My asterisk log is full of messages like this: [2011-03-06 19:01:15] DEBUG[20556] res_rtp_asterisk.c: Ignoring RTP 2833 Event: 009f. Not a DTMF Digit. [2011-03-06 19:01:20] DEBUG[20556] res_rtp_asterisk.c: Ignoring RTP 2833 Event: 009f. Not a DTMF Digit. [2011-03-06 19:01:25]

[asterisk-users] chan_sip.c: Failed to parse contact info

Hello! I have just upgraded to asterisk 1.8.2.1 and see some weird messages in log when client tries to register: [2011-01-19 00:52:47] WARNING[25624] chan_sip.c: Failed to parse contact info [2011-01-19 00:52:50] NOTICE[25624] chan_sip.c: Peer '0010101' is now UNREACHABLE! Last qualify: 105

Re: [asterisk-users] sip attack.. fail2ban not stopping attack

With asterisk 1.8+ it should be: failregex = NOTICE.* .*: Registration from '.*' failed for 'HOST(:[0-9]{1,5})?' - Wrong password NOTICE.* .*: Registration from '.*' failed for 'HOST(:[0-9]{1,5})?' - No matching peer found NOTICE.* .*: Registration from '.*' failed for

Re: [asterisk-users] asterisk realtime calling sip users

after some deep tracing it turned out to be a faulty router problem thanks. On Sun, Dec 26, 2010 at 9:38 AM, Sherwood McGowan sherwood.mcgo...@gmail.com wrote: On Sat, Dec 25, 2010 at 11:28 AM, Nick Ustinov nickusti...@gmail.com wrote: Hello We have recently upgraded to Realtime engine

[asterisk-users] SRTP unprotect: authentication failure

Hello! Ater several successful SRTP-enabled calls with SRTP set to Mandatory, asterisk starts to give the following warnings in Log: WARNING[13714] res_srtp.c: SRTP unprotect: authentication failure (continiously) and client hears no sound. After i restart the client program it works fine again

[asterisk-users] asterisk realtime calling sip users

Hello We have recently upgraded to Realtime engine (sip buddies and extensions) and now have problems with calling local SIP users. I have rtcachefriends=yes but tried with 'no' and it's even worse. (asterisk 1.8.1.1 + realtime mysql) Here's an example: User 1000 registers successfully and can

Re: [asterisk-users] sip attack.. fail2ban not stopping attack

Make sure you have dateformat=%F %T in logger.conf On Sun, Dec 26, 2010 at 1:04 AM, Dave George dgeo...@teletoneinc.com wrote: My server is being attached all day and fail2ban is not stopping the attack. I updated stamstamp to match fail2ban requirements. [2010-12-25 18:54:34]