I am sorry, I didn't show how put the pot in my last email, here it is,
dial-peer voice 150 voip description CCSi voip phone destination-pattern 9T progress_ind setup enable 3 session protocol sipv2 session target ipv4:216.236.160.11 codec g723r53
Answer to your question, without putting "isdn protocol-emulate network" I wasn't able to get PRI Layer 2 up.
Any other suggestion?
&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&* C.M. Rahman Jr. CTO 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: Richard [mailto:mypop3mail@yahoo.com] Sent: Friday, June 25, 2004 4:11 PM To: CM Rahman; serusers@lists.iptel.org Subject: RE: [Serusers] as5400 and ser
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:
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@lists.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:
Looking through your cisco config file, I am guessing your E1 are not Pri. Ami I correct? I am dealing with a
channelized
DS3 with T1 Pri. I will also share my config file after I can get the call routed. Currently I am getting this below. My
understanding
is there is something wrong in the call going from cisco to
Pri
trunk. Anybody can give me some clue, that will be great.
146.82.136.218:5060 -> 216.236.160.11:5060 SIP/2.0 503 Service Unavailable..Via:
SIP/2.0/UDP
216.236.160.11;branch=z9h G4bKc513.1c338976.0,SIP/2.0/UDP 65.70.207.66:8675..From: "pappusip@backup.c csi.com"
sip:pappusip@backup.ccsi.com;tag=c270cb2a9ab14343b72218adb808612
4;epid=c91b05026b..To:
sip:915125656553@backup.ccsi.com;tag=E8186070-487.
.Date: Tue, 15 Feb 2000 01:38:28 GMT..Call-ID: 9fef06800312431fbaa33d389f7d 3ac7@192.168.1.101..Server: Cisco-SIPGateway/IOS-12.x..CSeq: 1 INVITE..Allo w-Events: telephone-event..Content-Length: 0....
&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*
C.M. Rahman Jr. CTO 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: Stephen Kingham [mailto:Stephen.Kingham@aarnet.edu.au] Sent: Thursday, June 24, 2004 11:56 PM To: CM Rahman Cc: serusers@lists.iptel.org Subject: Re: [Serusers] as5400 and ser
Hi
Along with several other we are putting together a SER implementation Tutorial for the R&E sector.
We have a page up the the AS5300 and it may help you, also if anyone is interested in reviewing it?
http://www.aarnet.edu.au/events/conferences/2004/apan-questnet/sipworksh
op/uas/ciscoas5300.html
Regards
Stephen
CM Rahman wrote:
Anybody here using cisco as5400 for PSTN
termination? I am having some
problem with call routing. If there are such
person
will to help, please
=== message truncated ===
__________________________________ Do you Yahoo!? Read only the mail you want - Yahoo! Mail SpamGuard. http://promotions.yahoo.com/new_mail
CM Rahman wrote:
I am sorry, I didn't show how put the pot in my last email, here it is,
dial-peer voice 150 voip description CCSi voip phone destination-pattern 9T progress_ind setup enable 3 session protocol sipv2 session target ipv4:216.236.160.11 codec g723r53
Answer to your question, without putting "isdn protocol-emulate network" I wasn't able to get PRI Layer 2 up.
Yes. ISDN has a network side and a user side so that the layer 2 protocol Q921/lapb will work.
Most PABX want to be the user side.
Any other suggestion?
yes you have to have a pots dialpeer, the Cisco VoIP gateway requires at least one, I think maybe one for each E1 port.
Take a look at the template I have posted here: http://www.aarnet.edu.au/events/conferences/2004/apan-questnet/sipworkshop/u...
&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&* C.M. Rahman Jr. CTO 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: Richard [mailto:mypop3mail@yahoo.com] Sent: Friday, June 25, 2004 4:11 PM To: CM Rahman; serusers@lists.iptel.org Subject: RE: [Serusers] as5400 and ser
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:
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@lists.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:
Looking through your cisco config file, I am guessing your E1 are not Pri. Ami I correct? I am dealing with a
channelized
DS3 with T1 Pri. I will also share my config file after I can get the call routed. Currently I am getting this below. My
understanding
is there is something wrong in the call going from cisco to
Pri
trunk. Anybody can give me some clue, that will be great.
146.82.136.218:5060 -> 216.236.160.11:5060 SIP/2.0 503 Service Unavailable..Via:
SIP/2.0/UDP
216.236.160.11;branch=z9h G4bKc513.1c338976.0,SIP/2.0/UDP 65.70.207.66:8675..From: "pappusip@backup.c csi.com"
sip:pappusip@backup.ccsi.com;tag=c270cb2a9ab14343b72218adb808612
4;epid=c91b05026b..To:
sip:915125656553@backup.ccsi.com;tag=E8186070-487.
.Date: Tue, 15 Feb 2000 01:38:28 GMT..Call-ID: 9fef06800312431fbaa33d389f7d 3ac7@192.168.1.101..Server: Cisco-SIPGateway/IOS-12.x..CSeq: 1 INVITE..Allo w-Events: telephone-event..Content-Length: 0....
&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*
C.M. Rahman Jr. CTO 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: Stephen Kingham [mailto:Stephen.Kingham@aarnet.edu.au] Sent: Thursday, June 24, 2004 11:56 PM To: CM Rahman Cc: serusers@lists.iptel.org Subject: Re: [Serusers] as5400 and ser
Hi
Along with several other we are putting together a SER implementation Tutorial for the R&E sector.
We have a page up the the AS5300 and it may help you, also if anyone is interested in reviewing it?
http://www.aarnet.edu.au/events/conferences/2004/apan-questnet/sipworksh
op/uas/ciscoas5300.html
Regards
Stephen
CM Rahman wrote:
Anybody here using cisco as5400 for PSTN
termination? I am having some
problem with call routing. If there are such
person
will to help, please
=== message truncated ===
__________________________________ Do you Yahoo!? Read only the mail you want - Yahoo! Mail SpamGuard. http://promotions.yahoo.com/new_mail
Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
-
CM Rahman -
Stephen Kingham