similar to: warning : sip_xmit

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 Asterisk CLI is flooded with the following message : [Jun 25 21:24:57] WARNING[15174]: chan_sip.c:1851 __sip_xmit: sip_xmit of 0x409cd8f8 (len 519) to 109.236.137.138:1025 returned -1: Operation not permitted [Jun 25 21:25:01] WARNING[15174]: chan_sip.c:1851 __sip_xmit: sip_xmit of 0x409cd8f8 (len 519) to 109.236.137.138:1025 returned -1: Operation not permitted [Jun 25 21:25:05]

I'm having trouble setting up tls/srtp secure communications on my Asterisk server- I'm still rather new at working with Asterisk. I have enabled tls and encryption and I have csipsimple with tls build on the phone. I'm currently only testing one phone with this capability so far, and the rest still work in the current state. My logging looks like this with verbose turned up:

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",

I am integrating a new server (Asterisk 1.8.2.3, DAHDI 2.4.0) with VOIP only trunks, and this server only has soft phones. When I dial an extension and the phone is not registered, I don't get any ring or progress indications, and eventually the Dial() times out and drops into voicemail (as expected). CLI output: -- Executing [s at macro-StdExten:6] Dial("IAX2/barneveld-2036",

Hello Everyone - Well, I think I'm getting closer with the asterisk connection. This is my setup and I keep getting this error below in ,my /var/log/asterisk/messages file. I have opened 5060 port on the firewall box. I would this is Warning which I can ignore! But I see the connetcion coming but NO DIAL TONE on mt ata186 box sitting in my 192.168.200.x site! I'm using ATA186(cisco

Hello folks, since I started with asterisk 1.8.2 I got this messages in my console when finish a call. -- Executing [1610 at from-e1:1] Dial("SIP/xxx-00000027", "SIP/1610,60") in new stack == Using SIP RTP CoS mark 5 -- Called 1610 -- SIP/1610-00000028 is ringing -- SIP/1610-00000028 answered SIP/xxx-00000027 -- Locally bridging SIP/xxx-00000027 and

Following is my scenario to connect back to back PRI of two asterisk server. PRI cards are Sangoma A102D [Asterisk1]------------[PRI]-Cross Cable---------[Asterisk2] Asterisk1 ; Span 1 (MASTER) switchtype = national ; commonly referred to as NI2 context = from-pstn group = 0,24 echocancel = yes signalling = pri_net channel => 1-23 Asterisk2 ; Span 1 switchtype = national ; commonly

i'm using asterisk with tls but always get WARNING[17634]: chan_sip.c:3906 __sip_xmit: sip_xmit of 0x7fe394006590 (len 609) to 83.78.150.198:60709 returned -2: Success whats wrong there? Best Regards Jakob -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 473 bytes Desc: OpenPGP digital signature URL:

This morning, I received a very strange error message on the Asterisk console. The error occurs when I try to access iconnect WARNING[196621]: File chan_sip.c, Line 386 (__sip_xmit): sip_xmit of 0x80d0854 (len 649) to 213.137.73.178 returned -1: Bad file descriptor I also get this error when I try to reload: WARNING[16384]: File chan_sip.c, Line 5355 (reload_config): Unable to get IP address for

I have been trying to connect via sip and things don't seem to work. What do messages like this mean? Oct 9 00:33:57 WARNING[22849]: chan_sip.c:611 __sip_xmit: sip_xmit of 0x81ab834 (len 361) to 216.127.66.119 returned -1: Invalid argument Oct 9 00:33:58 WARNING[22849]: chan_sip.c:694 retrans_pkt: Maximum retries exceeded on call 000638cf3adb579455c0d20b2051ba1d@127.0.0.1 for seqno 102

Hi, After a few attempts, I've managed to grab the files from CVS and build it on a rh redora box I've setup especially for Asterisk. Firstly, we're new to the asterisk scene, so please excuse any "lame" questions which may follow.. We're a new voiptalk.org customer. We have purchased the voip phones (budgetone 102's) and set aside a little box to run Asterisk on.

When I try to make a call using the Mediatrix 1204 is showed on the CLI: -- Executing SetCIDNum("SIP/2009-4df1", "1111") in new stack -- Executing Dial("SIP/2009-4df1", "SIP/2217008@192.168.199.5") in new stack Aug 11 15:14:10 WARNING[1211108144]: chan_sip.c:590 __sip_xmit: sip_xmit of 0x81 40c5c (len 794) to 192.168.199.5 returned -1: Operation not

Hi. I cant make SIP calls from asterisk. When I start asterisk, I get the following message: What does it means?? Asterisk is not behind NAT or Firewall. ---------------------------------- [chan_sip.so] => (Session Initiation Protocol (SIP)) == Parsing '/etc/asterisk/sip.conf': Found Sep 16 09:52:33 WARNING[16384]: chan_sip.c:8477 reload_config: Unable to get IP address for

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

An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040108/748d21b3/attachment.htm -------------- next part -------------- Hello I downloaded latest CVS version and got following error messages while registeration. ( CVS Version Asterisk CVS-01/08/04-14:20:34) As a result IP Phone don't register with the Asterisk. Is it broken ? How can I

Hi all, My Asterisk problem today involves getting a SIP client on a private net to register with a server somewhere else on the Internet. This worked for me about a year ago no problem, but now I see an error message on the remote server every time the client attempts to connect (the server is running Debian lenny with Asterisk 1:1.4.21.2~dfsg-3). Here's an example: [Nov 11

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

I couldn't be happier with the simplicity of this - but it's a hack! Hi all, I'm currently using a SIP client (BT101) to connect via DSL to a remote instance of Asterisk. - Asterisk has a private IP behind my OFFICE router. - The SIP client has a private IP behind my HOME router. I'm doing this _without_ the use of STUN or proxy servers. Here's how it works: -

Hi. I have a x100p card installed and also asterisk, but I just dont get asterisk to register with my sip provider (FWD)... when I start asterisk using the following command I get the following messages (first, a lot of messages show up immediatly after starting up: I'read this is normal, then the CLI console comes out and this messages appear): NOTICE[229390]: chan_sip.c:3922