Hi, There seem to be some codec incompatibility. On *, you define alaw and you set ulaw on the Cisco.
Set both to same or add the other codec on (at least) one side. Try if that solve it Ex: Add "allow ulaw" on * after the "allow alaw" And / or Add "codec g711alaw" on Cisco above the "codec g711ulaw" If I remember correctly, Cisco parse the codecs according to their entry onder. Asterisk orders according to alphabetical order. If you do not need both codecs, set only one to simplify. Regards, Shaoul Jacobson Senior VoIP Consultant Tellink Tel : +32 3 201 96 36 Fax : +32 3 227 09 81 e-mail [EMAIL PROTECTED] -----Original Message----- From: Nathan Alberti [mailto:[EMAIL PROTECTED] Sent: mercredi 23 f�vrier 2005 0:45 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Call Manager Express Peer I have the following configuration and am obviously missing something small that is causing * not to work as expected. I have the following defined in sip.conf [ccme-in] type=peer host=10.0.9.1 context=devel_in disallow=all allow=alaw nat=no canreinvite=yes qualify=yes and [devel_in] is defined in extentions.conf However when I try to call via the dial peer I have configured on the cisco (below) I get : Feb 22 18:37:40 NOTICE[31486]: pbx.c:1318 pbx_extension_helper: Cannot find extension context 'default' Which is correct, meaning the context declaration is not being respected. ------ dial-peer voice 101 voip destination-pattern 10. session protocol sipv2 session target ipv4:10.0.0.133 dtmf-relay rtp-nte codec g711ulaw no vad ------- My bad or something else ?? TIA, Nathan. Here is a sip debug for that peer: Sending to 10.0.9.1 : 5060 (non-NAT) Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 10.0.9.1:19206 Found description format PCMU Found description format telephone-event Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) Looking for 101 in default Feb 22 18:44:25 NOTICE[31486]: pbx.c:1318 pbx_extension_helper: Cannot find extension context 'default' Reliably Transmitting (no NAT): SIP/2.0 404 Not Found Via: SIP/2.0/UDP 10.0.9.1:5060;branch=z9hG4bK3047A From: "Test Phone 1" <sip:[EMAIL PROTECTED]>;tag=17AFD44-10AD To: <sip:[EMAIL PROTECTED]>;tag=as3edc130d Call-ID: [EMAIL PROTECTED] CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:[EMAIL PROTECTED]> Content-Length: 0 to 10.0.9.1:5060 Destroying call '[EMAIL PROTECTED]' 11 headers, 0 lines Reliably Transmitting: OPTIONS sip:10.0.9.1 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.133:5060;branch=z9hG4bK2b06e290 From: "asterisk" <sip:[EMAIL PROTECTED]>;tag=as0a8b5343 To: <sip:10.0.9.1> Contact: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Tue, 22 Feb 2005 10:44:25 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 10.0.9.1:5060 Destroying call '[EMAIL PROTECTED]' _______________________________________________ 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 _______________________________________________ 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
