Once I send a call via messenger, I don't hear anything other side. But after a while it disconnect.
Here are the cisco config
****************************** controller T1 7/0:3 framing esf pri-group timeslots 1-24 description Prism Test
*************************************** interface Serial7/0:3:23 no ip address isdn switch-type primary-ni isdn protocol-emulate network isdn incoming-voice modem isdn T310 180000 no cdp enable !***************************************
dial-peer voice 150 voip description CCSi voip phone destination-pattern 9T session protocol sipv2 session target ipv4:216.236.160.11 codec g723r53
*****************************************
*Feb 15 16:18:09.720: ISDN Se7/0:3:23 Q931: Applying typeplan for sw-type 0xD is 0x2 0x1, Called num 5122200090 *Feb 15 16:18:09.720: ISDN Se7/0:3:23 Q931: TX -> SETUP pd = 8 callref = 0x002E Bearer Capability i = 0x8090A2 Standard = CCITT Transer Capability = Speech Transfer Mode = Circuit Transfer Rate = 64 kbit/s Channel ID i = 0xA98381 Exclusive, Channel 1 Called Party Number i = 0xA1, '5122200090' Plan:ISDN, Type:National *Feb 15 16:18:09.732: ISDN Se7/0:3:23 Q931: RX <- CALL_PROC pd = 8 callref = 0x802E Channel ID i = 0xA98381 Exclusive, Channel 1 *Feb 15 16:20:17.967: ISDN Se7/0:3:23 Q931: TX -> DISCONNECT pd = 8 callref = 0x002E Cause i = 0x8290 - Normal call clearing *Feb 15 16:20:17.995: ISDN Se7/0:3:23 Q931: RX <- RELEASE pd = 8 callref = 0x802E *Feb 15 16:20:17.995: ISDN Se7/0:3:23 Q931: TX -> RELEASE_COMP pd = 8 callref = 0x002E
&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&* C.M. Rahman Jr. CCNP, MCSE Security "Secure your self by securing your System" CompTI Security Plus Certified CCS Internet http://www.ccsi.com 13704 Research Blvd. Building O-Suite 4 Austin, TX 78750 Tel: 512-257-2274 Ex: 115
-----Original Message----- From: serusers-bounces@iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Richard Sent: Friday, June 25, 2004 3:27 AM To: serusers@lists.iptel.org Subject: RE: [Serusers] as5400 and ser
If you check this page,
http://www.cisco.com/en/US/tech/tk652/tk701/technologies_configuration_g uide_chapter09186a00800eadfa.html
PSTN error "63 Service or option unavailable" is mapped to sip error "503 Service or option unavailable" which is in the header of the message.
Also the page shows why IP phone or PSTN generates this and how proxy is supposed to do with it. Quote, "The SIP gateway generates this response if it is unable to process the request due to an overload or maintenance problem. Upon receiving this response, the gateway initiates a graceful call disconnect and clears the call. "
Look like a pstn config issue. Use "debug isdn q931", "debug isdn q921" and "term mon" for further debuging.
Cheers, Richard
--- CM Rahman cmrahman@ccsi.com wrote:
sip:pappusip@backup.ccsi.com;tag=c270cb2a9ab14343b72218adb808612
&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*
http://www.aarnet.edu.au/events/conferences/2004/apan-questnet/sipworksh
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Don't know why you have the following two lines, isdn protocol-emulate network isdn incoming-voice modem
Also you probably need a pots dial-peer...
Cisco web site has some configuration samples.
--- CM Rahman cmrahman@ccsi.com wrote:
&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*
http://www.cisco.com/en/US/tech/tk652/tk701/technologies_configuration_g
sip:pappusip@backup.ccsi.com;tag=c270cb2a9ab14343b72218adb808612
&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*
http://www.aarnet.edu.au/events/conferences/2004/apan-questnet/sipworksh
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-
CM Rahman -
Richard