similar to: chan_sip.c on debian testing - weird

Hi guys, I configure one Fedora Core Linux 5 for use with asterisk as gateway using Digium TE110P interconected in Alcantel 4100 I've set up it to register 100 voip numbers on my provider. All calls on Alcatel is send to asterisk. In some periods of day i receive this messages on asterisk console: Dec 12 17:49:30 NOTICE[11565]: chan_sip.c:11564 sip_poke_noanswer: Peer

Dne 13.8.2015 v 17:20 Rusty Newton napsal(a): > On Thu, Aug 13, 2015 at 3:54 AM, Marek ?ervenka <cervajs at fpf.slu.cz > <mailto:cervajs at fpf.slu.cz>> wrote: > > hello, > > is it possible simultaneously use chan_sip and chan_pjsip? > > if yes, can you recommend settings > > i'm thinking about > - chan_sip - for sip

Hello all, I downloaded and built the Asterisk v1.4.9 from the Debian Unstable repository on my Debian Etch GNU/Linux but when I checked the logs, I got some error messages from the chan_sip.c. You can find the logs below. # pwd /usr/src/debian/ # apt-get build-dep asterisk # exit $ cd /usr/src/debian/asterisk-1.4.9~dfsg/ $ debuild -us -uc ... ... ... - - - < s n i p > - - - Generating

I've got a test * server (hppbx) where I install CVS-HEAD as often as possible, with my extension registered to this, talking through IAX to our production server which then channels out to the PSTN. After completing a call just now, the following appeared on the CLI of hppbx (the 90xxxxxxxxxxx is a valid number, changed to protect the guilty): == Spawn extension (from-sip,

Greetings, Since the past week I've started receiving the following warnings on my asterisk servers (FreeBSD / CVS-HEAD). This warning manifests itself with x-lite/x-pro/eyebeam clients as well as sipura devices. All of them have qualify=yes in their settings. Jul 18 22:52:01 WARNING[73576]: chan_sip.c:939 __sip_xmit: sip_xmit of 0x8a3401c (len 483) to 195.x.y.28 returned -1: Address

Calling my home numbers has always worked. Till now. The Asterisk CLI show the following : [Dec 23 10:53:22] NOTICE[25159]: chan_sip.c:12640 handle_response_invite: Failed to authenticate on INVITE to '<sip:092xx90xx at 85.xx.xx.xx>;tag=as5b139383' And after restarting Asterisk, the CLI is flooded by : [Dec 23 11:11:06] WARNING[1468]: chan_sip.c:1817 __sip_xmit: sip_xmit of

Hello My friends, Today my asterisk stop working and i could see the following messags in /var/log/asterisk/messages at the time that asterisk stop working: [Feb 16 13:23:40] NOTICE[8230] chan_sip.c: Peer '324' is now Reachable. (2ms / 2000ms) [Feb 16 13:24:41] NOTICE[8230] chan_sip.c: Disconnecting call 'SIP/302-b720dd78' for lack of RTP activity in 301 seconds [Feb 16 13:25:54]

Hi, I have Asterisk 1.4.10 under LMCE (upgrade is not an option) and have this strange error appearing in full log : [Apr 16 14:35:48] NOTICE[10802] chan_sip.c: -- Registration for 'NUMBER at voip.siol' timed out, trying again (Attempt #22) [Apr 16 14:35:48] WARNING[10802] chan_sip.c: No such host: voip.siol [Apr 16 14:35:48] WARNING[10802] chan_sip.c: Probably a DNS error for

Hello everybody, I'm running Asterisk 1.2.24 on three servers which are configured almost identical. The servers use IAX to communicate between each other and SIP to communicate with the outside world through a Patton Smartnode 4960 gateway. One server has about 30 SIP phones registered, the other two servers have about 100 phones registered each. The "small" server runs fine

G'Day, I need from regarding using the X100P. Some are specific to Australia and I have been told in IRC there are already existing users in oz, so I would appreciate your input. 1) Is the X100P Australia telco approved? 2) I have setup asterisk with X100P and whenever I dial in I cannot get any callerid information. Is this standard behaviour? If not how have other australian users managed

Afternoon all, I'm trying to load Asterisk, however I am getting the following error: [skipping res_musiconhold.so] [chan_sip.so]May 22 18:42:24 WARNING[16384]: loader.c:240 ast_load_resource: /usr/lib/asterisk/modules/chan_sip.so: undefined symbol: ast_moh_stop May 22 18:42:24 WARNING[16384]: loader.c:421 load_modules: Loading module chan_sip.so failed! I've tried doing

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 [2011-01-19 00:53:03] VERBOSE[25624] chan_sip.c: -- Registered SIP '0010101' at

Hi, I have installed chan_sip on asterisk-1.0.3 / 5 (tried both, same result). My sip phone registers fine. But when dialing a number, I get: Feb 2 09:44:45 NOTICE[20380]: chan_sip.c:7296 handle_request: Unable to create/find channel ... Feb 2 09:44:52 WARNING[20380]: chan_sip.c:686 retrans_pkt: Maximum retries exceeded on call 384534305@192.168.1.20 for seqno 219 (Non-critical Response)

Hey Guys, Whenever i calling any extension i am getting following WARNING messages do you have any idea they coming from where? -Satish shirley*CLI> == Using SIP RTP CoS mark 5 -- Executing [7623 at from-sip:1] Macro("SIP/7527-00000008", "stdexten,7623,sip/7623&sip/7624") in new stack -- Executing [s at macro-stdexten:1] Dial("SIP/7527-00000008",

Hi guys I am writing a serial network driver. My aim is to bridge two ethernets across a serial bridge. Unfortunately the channel is noisy and slip and ppp are too unreliable. The driver appears to work by itself and allows two computers to ping each other across a null modem cable. The network driver code permits Linux OS ARPing support for ethernet and MAC addresses are manually assigned

Hi All, I'm running trixbox 2.0. The problem: a remote extension behind a NAT, can call other extensions, can call any other party, can call voicemail, will ring when rung, but when answered there is nothing and the dialling party continues to hear the ring tone. I'm getting this error in the logs: "That's odd... Got a response on a call we dont know about" I see

Hey all, I'm experiencing a problem with chan_sip on a multi-homed machine. The machine has 1 interface to the rest of the world and 1 interface on a local network. The local network has public IP-addresses, though, and the IP-addresses of both interfaces are reachable from the outside world, but by default, outgoing traffic from that machine to the outside world will have the IP-address of

Hello, We're attempting to use Asterisk for distributing calls via SIP in a large-scale speech recognition/VXML environment. We currently use DNS SRV with weights and priorities to instruct VoIP gateways (not Asterisk) to route calls to pools of servers. This works extremely well and provides for load balancing, fail-over, and by setting the TTL low (several minutes) we can easily take

Hi All, how are you? I would like to know from you if the problem can be below is a BUG of the asterisk-1.4.21. I did an upgrade version of asterisk-1.2.18 for the version of asterisk-1.4.21 and now, when users try to sip friend outgoing calls through Polycom IP 330 appliances can not be the traditional way or with the telephone handset in his hand and digit dialing digit to receive the following

This is a great news, thank you. I have open the issue, https://issues.asterisk.org/jira/browse/ASTERISK-26177 and added the relevant files, let me know if you need more info. Leandro 2016-07-06 21:46 GMT+02:00 Joshua Colp <jcolp at digium.com>: > Leandro Dardini wrote: > >> Hello, >> I'd like to know if anyone of you is finding my same problems using any >>