hello. i have a problem.
i can work around it, but i would lose functionnality and i dont want to.
i have a avaya cs1000, and multiple asterisk.
the avaya has 6XXX and 3XXX (in general).
the asterisk has the 2XXX numbers (again, in general)
if i take a 2xxx, call 6xxx, wich in turn has a forward call back to a 2xxx number, it fails.
if i have a 6xxx call that forward calls to a 2xxx voicemail:
calls from 6xxx and 3xxx work, 2xxx calls fail.
fail message is
Using INVITE request as basis request - ad3812a8-1e0310ac-13c4-55013-549b9f-2c423a6f-549b9f
Found peer ‘2997’ for ‘2997’ from 172.16.3.153:5060
[2016-01-08 14:51:04] WARNING[23070]: chan_sip.c:15354 check_auth: username mismatch, have <2997>, digest has <MO_ST-ANNE>
[2016-01-08 14:51:04] NOTICE[23070]: chan_sip.c:23900 handle_request_invite: Failed to authenticate device “hugo tab” <sip:2997;phone-context=UnknownUnknown@cssh.qc.ca686b28-1e0310ac-13c4-55013-549b9f-3535ff85-549b9f
outbound route is set as internal, since it points to internal numbers and i need to keep internal CID.
it was as is before the trouble too and working fine.
now, the work around is to treat it as an external route, losing the 2997. when it comes back as an external number, asterisk dont see it as an internal peer and its good. but its not really an usable option…
trunk is a sip trunk:
[nortel]
disallow=all
type=peer
qualify=yes
port=5060
nat=no
host=172.16.3.153
context=from-trunk-sip-nortel
canreinvinte=no
allow=ulaw
allow=alaw
allow=gsm
i’m sure it was working before an upgrade, so it must be something in the configs.
its a production server (call it PS) that has been updraded to the same version as another one (call it BS). PS was an old asterisk/freepbx, has been upgraded to same version as BS, PS was backup’d and restored to BS, in effect transfering all the configs from one to the other.
system has been running for a while. old PS was up 4 years, new BS was up 2 months.
here is the sip set debug which contain all the usable info:
<— SIP read from UDP:172.16.3.153:5060 —>
INVITE sip:2993;phone-context=cdp.udp@cssh.qc.ca;user=phone SIP/2.0
Record-Route: sip:7917e1d2@172.16.3.153;transport=udp;lr
Record-Route: sip:172.16.3.152:15060;lr;sap=752120364*1*016asm-callprocessing.sar-1744629844~1452282664473~590664232~1
Record-Route: sip:7917e1d2@172.16.3.153;transport=tcp;lr
From: “hugo tab” sip:2997;phone-context=UnknownUnknown@cssh.qc.ca;user=phone;tag=ad686b28-1e0310ac-13c4-55013-549b9f-3535ff85-549b9f
To: sip:2993;phone-context=cdp.udp@cssh.qc.ca;user=phone
Call-ID: ad3812a8-1e0310ac-13c4-55013-549b9f-2c423a6f-549b9f
CSeq: 2 INVITE
Via: SIP/2.0/UDP 172.16.3.153;rport;branch=z9hG4bKAC100398FFFFFFFFC9C0854B0944636-AP;ft=172.16.3.153~13c4
Via: SIP/2.0/UDP 172.16.3.152:15070;branch=z9hG4bKAC100398FFFFFFFFC9C0854B0944636
Via: SIP/2.0/UDP 172.16.3.152:15070;branch=z9hG4bKAC100398FFFFFFFFC9C0854B1944634
Via: SIP/2.0/UDP 172.16.3.152:15070;branch=z9hG4bKAC100398FFFFFFFFC9C0854B1944633
Via: SIP/2.0/TCP 172.16.3.153;branch=z9hG4bK-549ba0-4a7fe906-19d6de68-AP;ft=648
Via: SIP/2.0/TCP 172.16.3.30:5060;branch=z9hG4bK-549ba0-4a7fe906-19d6de68
Supported: x-nortel-sipvc, replaces
User-Agent: Nortel CS1000 SIP GW release_7.0 version_ssLinux-7.50.17 AVAYA-SM-6.1.7.0.617012
P-Asserted-Identity: “hugo tab” sip:2997;phone-context=UnknownUnknown@cssh.qc.ca;user=phone
Privacy: none
History-Info: sip:6213;phone-context=cdp.udp@cssh.qc.ca;user=phone?reason=sip%3Bcause%3D302%3Btext%3D"Moved%20Temporarily";index=1,sip:2993;phone-context=cdp.udhone;index=2
Alert-Info: cid:external@cssh.qc.ca
Contact: sip:2997;phone-context=UnknownUnknown@cssh.qc.ca:5060;maddr=172.16.3.30;transport=tcp;user=phone
Authorization: Digest username=“MO_ST-ANNE”,realm=“asterisk”,nonce=“74e0c343”,uri="sip:2993;phone-context=cdp.udp@cssh.qc.ca;user=phone",response="ae95d6b54afec9375bb1ithm=MD5
Allow: INVITE,ACK,BYE,REGISTER,REFER,NOTIFY,CANCEL,OPTIONS,INFO,SUBSCRIBE
Content-Type: multipart/mixed;boundary=unique-boundary-1
Content-Length: 781
Route: sip:10.24.11.217;lr;phase=terminating
P-Location: SM;origlocname=“CSSH”;termlocname="CSSH"
Max-Forwards: 66
the History tag show why its coming back. 6213, moved temporarily (call forwarded) to 2993, but the message is kind of all distorted…
i use the legacy_useroption_parsing = yes, because the avaya add ";phone-context=cdp.udp@cssh.qc.ca;user=phone" to the number, where before i used a little patch in my context:
;exten => _X.,1,Set(GROUP()=OUT_1)
;exten => _X.,2,Goto(from-trunk,${EXTEN:0:-22},1)
to remove the message.
anyone has an idea how to resolve this? i can work on it on the weekends, but the week days, between 6 and 18 its pretty busy…
i really need help here. thanks a lot!
hugo