similar to: chan_sip errors on CVS HEAD

Just put a new phone in place with the latest firmware from Cisco. This is the first SPA501G we have with this firmware. In the Asterisk CLI we are now seeing the error message below about once every second. When we unplug the phone, the messages quit. NOTICE[15539]: chan_sip.c:9602 copy_header: No field 'CSeq' present to copy Thanks in advance for any assistance on this.

I have managed to get spandsp working, and if I dial a specific extension I can receive faxes. WhooHoo. However, I was wanting to use the "fax detect" option in order to allow individuals to receive faxes, but can't get that to work. Given the following extensions (mainly copied from examples on the wiki), why is the call simply passed onto the sip device rather than being

hi! anyone try siemens optipoint 400 economy SIP phone with * ? -- http://www.siemens.com/Daten/siecom/HQ/ICN/Internet/Enterprise_Networks/WORKAREA/skuch_c/templatedata/English/document/binary/a31002-h1000-a250-2-7629.pdf Thomas

Hello, After checking out CVS HEAD from yesterday (for those new PLC/Jitterbuffer patches), I was affected by bug 3795 with my Polycom IP600's. After seing it resolved as of this morning (thanks Mark), I decided to try again... I can answer incoming calls. No problem there. Putting calls on hold, however, results in my Polycom IP600 indicating the call on hold, but the caller does

....I THINK. When dialing 1+10 digits, I occasionally get a telco message "You must first dial a 1....". When I look at the console, the number is being sent to the ZAP channel properly. We're talking about a couple of POTS lines on a TDM400P. I'm thinking that it may be starting the dial too early after coming off-hook because I can just redial and have it work (or not)

hi new user here cant seem to get fwd running, got asterisk from download site as tarball, did the readln and openssl start. Also configured the sip.conf and extensions.conf but an error with the chan_sip.c shows up? any ideas...somebody...anybody! thanx jai

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

Hi, i am trying to install asterisk newer version on the Elastix machine, but while installing the chan_sip,c module is not building while make. when i see in make menuselect options it showing "XXX" -- extended , please let me know how to enable it and make build chan_sip module. -- Upendra -------------- next part -------------- An HTML attachment was scrubbed... URL:

All, I've found problems in my setup with the latest couple of revisions (1.135/1.136) of asterisk/channels/chan_sip.c In my setup I have a RH9 asterisk server, AS5300 (single E1 to PSTN) and a dozen 7940's, everything is in the same VLAN and only running SIP. Outbound calls work fine: 7940 -SIP-> Asterisk -SIP-> AS5300 But inbound calls fail, I see the initial INVITE from the

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, I'd like to know if anyone of you is finding my same problems using any recent asterisk version, after 13.7 / 13.8 with chan_sip. If I use any recent asterisk version, after just few seconds asterisk completely locks up, stopping processing SIP/UDP packets. Nothing is written in the asterisk log, but if I run "netstat -nap | grep 5060" I see the UDP buffer filled up. If I

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",

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

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 hardphones/softphones (sip udp 5060) - chan_pjsip - for webrtc -- --------------------------------------- Marek Cervenka =======================================

I am having real trouble getting started. A definitive "hello world" is certainly missing from the official site and the ones out there are dated or broken. I am beginning to think something went wrong with the install. It was a fresh install of an Ubuntu server, and a fresh install of 13.7.0 - Should be Okay no? A question. Am I expecting to find chan_sip.so in

Am I missing something here? I see if (needvideo) { /* only if video response is appropriate */ add_line(resp, m_video->str); add_line(resp, a_video->str); add_line(resp, hold); /* Repeat hold for the video stream */ } else if (p->offered_media[SDP_VIDEO].offered) { snprintf(dummy_answer,

asterisk-users-bounces@lists.digium.com wrote: > Is there anyone else with the same problem? Yes, we've seen the same problem. We have found a work around, but I'm unable to to look into it today. -- Andreas Sikkema Rits tele.com Van Vollenhovenstraat 3 3016 BE Rotterdam t: +31 (0)10 2245544 f: +31 (0)10 2245540

hi listmembers, please test my new patch to chan_sip.c which is to make call pickup on the snom phones (and maybe other phones that support 'INVITE/Replaces') work and make comments in the bugtracker http://bugs.digium.com/view.php?id=3644 so it can make its way into the cvs. this patch sipsubscribe-20050715.rev779.txt enables: * monitoring of other lines (using the 'hint'