I've found the problem.
Sip read: SIP/2.0 400 Bad Request Via: SIP/2.0/UDP 172.31.30.3:5060;branch=z9hG4bK658d8eed;rport From: ""Nik Martin IAX" <sip:[EMAIL PROTECTED]>;tag=as40e79563 <<<<<<<<<<<<<<<<< To: <sip:[EMAIL PROTECTED]:5060;user=phone> Call-ID: [EMAIL PROTECTED] Date: Wed, 03 Nov 2004 15:21:52 GMT Warning: 399 Bad Request - 'Malformed/Missing FROM: field' <<<<<<<<<<<<<<<<<< CSeq: 102 INVITE Content-Length: 0 Prior to upgrading, I had caller ID strings in IAX.Conf like: Callerid="Nik Martin <105>" Now, evedintly these must look like: Callerid "Nik Martin" <105> This was probably an error on my part, but prior to upgrading, it seemed to work fine. I don't remember if callerd id was actually being sent from IAX phones or not, but it worked. -----Original Message----- From: Nik Martin [mailto:[EMAIL PROTECTED] Sent: Wednesday, November 03, 2004 9:29 AM To: '[EMAIL PROTECTED]' Subject: IAXys or IAX Softphones cannot call SIP phones I recently upgraded to the latest asterisk, and everything was working fine. Just recently, my softphones or IAXys cannot call any SIP phones. This happens whether the IAX calls originate on my LAN, or outside the network. The Cisco SIP phones can call the IAX phones fine. I thought it was codec related, but in iax.conf, I've disallowed everything but alaw. Here's the SIP debug. Thanks for any insight. This is a firefly (latest thirdparty release) softphone calling a SIP phone (Cisco 7960): pbxMobile*CLI> sip debug SIP Debugging Enabled -- Accepting AUTHENTICATED call from 172.31.30.20, requested format = 1024, actual format = 4 -- Executing Dial("IAX2/[EMAIL PROTECTED]:4569/8", "SIP/nmartin|20|tT") in new stack We're at 172.31.30.3 port 11950 Answering/Requesting with root capability 4 Answering with preferred capability 0x8(ALAW) 12 headers, 9 lines Reliably Transmitting: INVITE sip:[EMAIL PROTECTED]:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 172.31.30.3:5060;branch=z9hG4bK658d8eed;rport From: ""Nik Martin IAX" <sip:[EMAIL PROTECTED]>;tag=as40e79563 To: <sip:[EMAIL PROTECTED]:5060;user=phone> Contact: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Wed, 03 Nov 2004 15:16:25 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 182 v=0 o=root 15333 15333 IN IP4 172.31.30.3 s=session c=IN IP4 172.31.30.3 t=0 0 m=audio 11950 RTP/AVP 0 8 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - (NAT) to 172.31.30.7:5060 -- Called nmartin pbxMobile*CLI> Sip read: SIP/2.0 400 Bad Request Via: SIP/2.0/UDP 172.31.30.3:5060;branch=z9hG4bK658d8eed;rport From: ""Nik Martin IAX" <sip:[EMAIL PROTECTED]>;tag=as40e79563 To: <sip:[EMAIL PROTECTED]:5060;user=phone> Call-ID: [EMAIL PROTECTED] Date: Wed, 03 Nov 2004 15:21:52 GMT Warning: 399 Bad Request - 'Malformed/Missing FROM: field' CSeq: 102 INVITE Content-Length: 0 _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
