[Asterisk-Users] getting callerid from spa3k to asterisk
ok, with a good pointer from Chris Stenton [EMAIL PROTECTED], i found the problem. if i have two sip contexts for my spa3k, on inbound and one outbound, e.g. [spa3k-out] type=peer auth=md5 secret=pfui username=outpass fromuser=outpass host=spa3k.bogus.com port=5061 nat=no canreinvite=yes context=ext-in42 [spa3k-in] type=friend host=dynamic port=5061 auth=md5 secret=pfui qualify=1000 canreinvite=yes context=ext-in42 and the spa3k's PSTN / Subscriber Information / User ID: = spack-in, the incoming connection from spa3k to * is being routed to the spa3k-out context, not the spa3-in context. see appended. i suspect this is a bug in * 1.0.1. so, until the problem is diagnosed, how do i work around it. as the spa3k is registered, i tried to remove the spa3k-out context entirely. callerid now works. yes! but ... if i try to place an outbound call using the spa3k-in context, the call is sent to the spa3k, but it just gives me the pstn's dialtone, and does not dial the number. my spa3k config is in http://rip.psg.com/~randy/spa3k.html. so how do i place a call out the spa3k pstn without a separate outbound context? randy --- Sip read: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 198.180.150.195:5061;branch=z9hG4bK-69580ec1 From: CallerName sip:[EMAIL PROTECTED];tag=25aee11517d597a1o1 To: sip:[EMAIL PROTECTED] Remote-Party-ID: CallerName sip:[EMAIL PROTECTED];screen=yes;party=calling Call-ID: [EMAIL PROTECTED] CSeq: 101 INVITE Max-Forwards: 70 Contact: biwa 0431 sip:[EMAIL PROTECTED]:5061 Expires: 240 User-Agent: Sipura/SPA3000-2.0.11(GWa) Content-Length: 428 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura Content-Type: application/sdp v=0 o=- 8805171 8805171 IN IP4 198.180.150.195 s=- c=IN IP4 198.180.150.195 t=0 0 m=audio 16396 RTP/AVP 0 2 4 8 18 96 97 98 100 101 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729a/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:100 NSE/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv 15 headers, 19 lines Using latest request as basis request Sending to 198.180.150.195 : 5061 (non-NAT) Found RTP audio format 0 Found RTP audio format 2 Found RTP audio format 4 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 96 Found RTP audio format 97 Found RTP audio format 98 Found RTP audio format 100 Found RTP audio format 101 Peer audio RTP is at port 198.180.150.195:16396 Found description format PCMU Found description format G726-32 Found description format G723 Found description format PCMA Found description format G729a Found description format G726-40 Found description format G726-24 Found description format G726-16 Found description format NSE Found description format telephone-event Capabilities: us - 0xe(GSM|ULAW|ALAW), peer - audio=0x51d(G723|ULAW|ALAW|G726|G729A|ILBC)/video=0x0(EMPTY), combined - 0xc(ULAW|ALAW) Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 0x1(G723) Found peer 'spa3k-out' ___ 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
Re: [Asterisk-Users] getting callerid from spa3k to asterisk
Maybe this will be of help. http://voip-forum.tmcnet.com/voip-forum/forum/forum_posts.asp?TID=1707PN=0TPN=1 - Original Message - From: Randy Bush [EMAIL PROTECTED] To: splatters [EMAIL PROTECTED] Sent: Friday, November 12, 2004 9:13 PM Subject: [Asterisk-Users] getting callerid from spa3k to asterisk ok, with a good pointer from Chris Stenton [EMAIL PROTECTED], i found the problem. if i have two sip contexts for my spa3k, on inbound and one outbound, e.g. [spa3k-out] type=peer auth=md5 secret=pfui username=outpass fromuser=outpass host=spa3k.bogus.com port=5061 nat=no canreinvite=yes context=ext-in42 [spa3k-in] type=friend host=dynamic port=5061 auth=md5 secret=pfui qualify=1000 canreinvite=yes context=ext-in42 and the spa3k's PSTN / Subscriber Information / User ID: = spack-in, the incoming connection from spa3k to * is being routed to the spa3k-out context, not the spa3-in context. see appended. i suspect this is a bug in * 1.0.1. so, until the problem is diagnosed, how do i work around it. as the spa3k is registered, i tried to remove the spa3k-out context entirely. callerid now works. yes! but ... if i try to place an outbound call using the spa3k-in context, the call is sent to the spa3k, but it just gives me the pstn's dialtone, and does not dial the number. my spa3k config is in http://rip.psg.com/~randy/spa3k.html. so how do i place a call out the spa3k pstn without a separate outbound context? randy --- Sip read: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 198.180.150.195:5061;branch=z9hG4bK-69580ec1 From: CallerName sip:[EMAIL PROTECTED];tag=25aee11517d597a1o1 To: sip:[EMAIL PROTECTED] Remote-Party-ID: CallerName sip:[EMAIL PROTECTED];screen=yes;party=calling Call-ID: [EMAIL PROTECTED] CSeq: 101 INVITE Max-Forwards: 70 Contact: biwa 0431 sip:[EMAIL PROTECTED]:5061 Expires: 240 User-Agent: Sipura/SPA3000-2.0.11(GWa) Content-Length: 428 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura Content-Type: application/sdp v=0 o=- 8805171 8805171 IN IP4 198.180.150.195 s=- c=IN IP4 198.180.150.195 t=0 0 m=audio 16396 RTP/AVP 0 2 4 8 18 96 97 98 100 101 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729a/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:100 NSE/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv 15 headers, 19 lines Using latest request as basis request Sending to 198.180.150.195 : 5061 (non-NAT) Found RTP audio format 0 Found RTP audio format 2 Found RTP audio format 4 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 96 Found RTP audio format 97 Found RTP audio format 98 Found RTP audio format 100 Found RTP audio format 101 Peer audio RTP is at port 198.180.150.195:16396 Found description format PCMU Found description format G726-32 Found description format G723 Found description format PCMA Found description format G729a Found description format G726-40 Found description format G726-24 Found description format G726-16 Found description format NSE Found description format telephone-event Capabilities: us - 0xe(GSM|ULAW|ALAW), peer - audio=0x51d(G723|ULAW|ALAW|G726|G729A|ILBC)/video=0x0(EMPTY), combined - 0xc(ULAW|ALAW) Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 0x1(G723) Found peer 'spa3k-out' ___ 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
[Asterisk-Users] getting callerid from spa3k to asterisk
i am still dying on this one, and my critical user, my fiance'e,
is giving me hell over it on my home test environment; even my
daytime job, for which i am prototyping, is more patient. :-)
i can not get caller-id from a call coming in to the spa3k pstn
to asterisk. fwiw, this used to work with older * and spa3k
versions, but of course it could be something i did to configs.
essentially, if i tell the spa3k to pass callerid to *, the sip
session gets rejected by *. since no one seemed to like to see
ethereal output, i have posted * sip debug form of the sessions.
does anyone have their spa3k and * config working that i could
look at? or, if you can shoot the bug, i'll pay you US$100 by
paypal or whatever.
the spa3k configiuration
http://rip.psg.com/~randy/spa3k.html
sip debug with spa3k config set to
PSTN / PSTN-To-VoIP Gateway Setup / PSTN CID For VoIP CID: = NO
call accepted ok, but no callerid received by asterisk
http://rip.psg.com/~randy/debug-0.txt
sip debug with spa3k config set to
PSTN / PSTN-To-VoIP Gateway Setup / PSTN CID For VoIP CID: = YES
call rejected by asterisk
http://rip.psg.com/~randy/debug-1.txt
sip.conf entry
[spa3k-in]
type=friend ; user fails to register
host=dynamic
port=5061
auth=md5
secret=dontbesilly
qualify=1000
dtmfmode=rfc2833
canreinvite=yes
context=ext-in42
extensions.conf for the incoming
[ext-in42]
exten = _X.,1,NoOp(ext-in42 cid=${CALLERIDNUM})
exten = _X.,2,SetVar(areacode=206)
exten = _X.,3,SetVar(mailbox=1)
exten = _X.,4,GoTo(ext-common,s,1)
[ext-common]
exten = s,1,NoOp(ext-common cid=${CALLERIDNUM})
exten = s,2,Background(zz-who-common)
exten = i,1,Hangup()
exten = t,1,GoTo(ext-common,s,1)
include = speeddials
include = extensions
include = conferences
include = applications
randy
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Re: [Asterisk-Users] getting callerid from spa3k to asterisk
Randy Bush wrote: essentially, if i tell the spa3k to pass callerid to *, the sip session gets rejected by *. since no one seemed to like to see ethereal output, i have posted * sip debug form of the sessions. You could maybe look at the autocreatepeer option for sip.conf - just a wild guess so YMMV. BTW: Sorry didn't click through to your traces so this might be pointless! :-) Think that's enough disclaimers! -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ 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
