Richard Bruce
2012-11-01 02:39:58 UTC
I am attempting to set up a Digium G100 T1 Gateway on a SipXecs 4.4.0x
server. The G100 seems to have issues with REFER as I've experienced with
some other gateways. I was able to get around a transfer issue by creating
route to send transferred extensions back to the SipXecs server. I still
have an issue with the Call Park. I was able to create the same kind of
route to transfer calls to the Call Park number, but when I try to retrieve
the parked call I have similar issues to another recent post. I hear a
quick cut in the On Hold Music on the callers end like it is trying to
transfer, but then goes back to hold and I cannot retrieve the call.
This is a capture of the SIP info:
Oct 31 09:16:08 G100-59-be-97 asterisk[3621]: VERBOSE[4022]:
chan_sip.c:10197 in set_destination: set_destination: set destination to
172.18.10.10:5060
Oct 31 09:16:08 G100-59-be-97 asterisk[3621]: VERBOSE[4022]: chan_sip.c:4249
in send_request: Reliably Transmitting (NAT) to 172.18.10.10:5060: NOTIFY
sip:***@172.18.10.10:5120;transport=tcp SIP/2.0^M Via: SIP/2.0/UDP
172.18.10.20:5060;branch=z9hG4bK75d8a39d;rport^M Route:
<sip:172.18.10.10:5060;lr;sipXecs-CallDest=PARK;sipXecs-rs=%2Aauth%7E.%2Afro
m%7EYXMyODIxMmJjOA%60%60.900_ntap%2Aid%7EMjk5ODYtMTI1%21e116f5ad21ca3e7e53cb
51f43a5569ef;x-sipX-done>^M Max-Forwards: 70^M From:
"3178177001"<sip:***@172.18.10.20>;tag=as28212bc8^M To:
<sip:***@172.18.10.10>;tag=267AhV^M Contact: <sip:***@172.18.10.20:5060>^M
Call-ID: ***@172.18.10.20:5060^M CSeq: 103
NOTIFY^M User-Agent: Digium Gateway^M Event: refer;id=3^M
Subscription-state: terminated;reason=noresource^M Content-Type:
message/sipfrag;version=2.0^M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE,
REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH^M Supported: replaces, timer^M
Content-Length: 49^M ^M SIP/2.0 481 Call leg/transaction does not exist^M
---
Oct 31 09:16:08 G100-59-be-97 asterisk[3621]: VERBOSE[4022]:
chan_sip.c:25869 in handle_request_do: <--- SIP read from
UDP:172.18.10.10:5060 ---> SIP/2.0 200 OK^M From:
"3178177001"<sip:***@172.18.10.20>;tag=as28212bc8^M To:
<sip:***@172.18.10.10>;tag=267AhV^M Call-Id:
***@172.18.10.20:5060^M Cseq: 103 NOTIFY^M
Contact: <sip:***@172.18.10.10:5120;transport=tcp>^M Via: SIP/2.0/UDP
172.18.10.20:5060;branch=z9hG4bK75d8a39d;rport=5060;id=29986-125^M Date:
Wed, 31 Oct 2012 13:15:42 GMT^M Allow: INVITE, ACK, CANCEL, BYE, REFER,
OPTIONS, NOTIFY, SUBSCRIBE^M User-Agent: sipXecs/4.4.0 sipXecs/park
(Linux)^M Accept-Language: en^M Supported: replaces^M Content-Length: 0^M ^M
<------------->
Oct 31 09:16:08 G100-59-be
Looking for any input.
If this gateway will work, I would be glad to document any info for the
WIKI.
Thanks,
Richard Bruce
Dimensional Communications
7915 S. Emerson Ave, Suite 131
Indianapolis, IN 46237
(317) 215-4199- office
(317) 946-1899 - cell
server. The G100 seems to have issues with REFER as I've experienced with
some other gateways. I was able to get around a transfer issue by creating
route to send transferred extensions back to the SipXecs server. I still
have an issue with the Call Park. I was able to create the same kind of
route to transfer calls to the Call Park number, but when I try to retrieve
the parked call I have similar issues to another recent post. I hear a
quick cut in the On Hold Music on the callers end like it is trying to
transfer, but then goes back to hold and I cannot retrieve the call.
This is a capture of the SIP info:
Oct 31 09:16:08 G100-59-be-97 asterisk[3621]: VERBOSE[4022]:
chan_sip.c:10197 in set_destination: set_destination: set destination to
172.18.10.10:5060
Oct 31 09:16:08 G100-59-be-97 asterisk[3621]: VERBOSE[4022]: chan_sip.c:4249
in send_request: Reliably Transmitting (NAT) to 172.18.10.10:5060: NOTIFY
sip:***@172.18.10.10:5120;transport=tcp SIP/2.0^M Via: SIP/2.0/UDP
172.18.10.20:5060;branch=z9hG4bK75d8a39d;rport^M Route:
<sip:172.18.10.10:5060;lr;sipXecs-CallDest=PARK;sipXecs-rs=%2Aauth%7E.%2Afro
m%7EYXMyODIxMmJjOA%60%60.900_ntap%2Aid%7EMjk5ODYtMTI1%21e116f5ad21ca3e7e53cb
51f43a5569ef;x-sipX-done>^M Max-Forwards: 70^M From:
"3178177001"<sip:***@172.18.10.20>;tag=as28212bc8^M To:
<sip:***@172.18.10.10>;tag=267AhV^M Contact: <sip:***@172.18.10.20:5060>^M
Call-ID: ***@172.18.10.20:5060^M CSeq: 103
NOTIFY^M User-Agent: Digium Gateway^M Event: refer;id=3^M
Subscription-state: terminated;reason=noresource^M Content-Type:
message/sipfrag;version=2.0^M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE,
REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH^M Supported: replaces, timer^M
Content-Length: 49^M ^M SIP/2.0 481 Call leg/transaction does not exist^M
---
Oct 31 09:16:08 G100-59-be-97 asterisk[3621]: VERBOSE[4022]:
chan_sip.c:25869 in handle_request_do: <--- SIP read from
UDP:172.18.10.10:5060 ---> SIP/2.0 200 OK^M From:
"3178177001"<sip:***@172.18.10.20>;tag=as28212bc8^M To:
<sip:***@172.18.10.10>;tag=267AhV^M Call-Id:
***@172.18.10.20:5060^M Cseq: 103 NOTIFY^M
Contact: <sip:***@172.18.10.10:5120;transport=tcp>^M Via: SIP/2.0/UDP
172.18.10.20:5060;branch=z9hG4bK75d8a39d;rport=5060;id=29986-125^M Date:
Wed, 31 Oct 2012 13:15:42 GMT^M Allow: INVITE, ACK, CANCEL, BYE, REFER,
OPTIONS, NOTIFY, SUBSCRIBE^M User-Agent: sipXecs/4.4.0 sipXecs/park
(Linux)^M Accept-Language: en^M Supported: replaces^M Content-Length: 0^M ^M
<------------->
Oct 31 09:16:08 G100-59-be
Looking for any input.
If this gateway will work, I would be glad to document any info for the
WIKI.
Thanks,
Richard Bruce
Dimensional Communications
7915 S. Emerson Ave, Suite 131
Indianapolis, IN 46237
(317) 215-4199- office
(317) 946-1899 - cell