-----Original Message----- From: Adams, Gavin Sent: Monday, August 04, 2003 6:10 PM To: 'asterisk-users@lists.digium.com' Subject: Cisco 7960, SIP, NAT, Voicemal Hey all, I've got a couple 79xx phones working peer-to-peer and am now trying to work on the voice mail. In extensions.conf: [ATL] exten => 4001,1,Dial(SIP/gadams)|10 exten => 4001,2,Voicemail,u4001 exten => 4001,102,Voicemail,b4001 and the corresponding sip.conf: [gadams] type=friend username=gadams secret=****** context=ATL host=dynamic canreinvite=no nat=yes mailbox=4001 When this phone is dialed, it doesn't roll over to VM after 10 seconds but continues to ring. If the calling party hangs up, the phone continues to ring. However, as a test I changed the |10 to a |10t. At that point dialing 4001 did indeed roll over to voicemail, but it happened immediately. Also, I'm getting the following message during the dial: WARNING[1133735216]: File chan_sip.c, Line 417 (retrans_pkt): Maximum retries exceeded on call 71817d20169869e82ee3d897699393e3@63.111.7.161 for seqno 102 (Request) Which is tied to the call in question. Any clues? --- Gavin
Try this: exten => 4001,1,Dial(SIP/gadams,10,r) I don't know how the syntax you've specified will behave; maybe it will work, but it's not any format I've used. Try the syntax above for your Dial line and see if it results in different actions. Plus, I'm slightly confused as to your dialplan. When a call comes in from your 7960, it is passed to context "ATL". Now, when you call your 7960, where is that call going? Are you passing your inbound and outbound calls both to context "ATL"? If not, then you're missing an important part of the debug (and perhaps having a conceptual problem with how call flow works.) JT>-----Original Message----- >From: Adams, Gavin >Sent: Monday, August 04, 2003 6:10 PM >To: 'asterisk-users@lists.digium.com' >Subject: Cisco 7960, SIP, NAT, Voicemal > >Hey all, > >I've got a couple 79xx phones working peer-to-peer and am now trying to >work on the voice mail. > >In extensions.conf: > >[ATL] >exten => 4001,1,Dial(SIP/gadams)|10 >exten => 4001,2,Voicemail,u4001 >exten => 4001,102,Voicemail,b4001 > >and the corresponding sip.conf: > >[gadams] >type=friend >username=gadams >secret=****** >context=ATL >host=dynamic >canreinvite=no >nat=yes >mailbox=4001 > >When this phone is dialed, it doesn't roll over to VM after 10 seconds >but continues to ring. If the calling party hangs up, the phone >continues to ring. > >However, as a test I changed the |10 to a |10t. At that point dialing >4001 did indeed roll over to voicemail, but it happened immediately. >Also, I'm getting the following message during the dial: > >WARNING[1133735216]: File chan_sip.c, Line 417 (retrans_pkt): Maximum >retries exceeded on call 71817d20169869e82ee3d897699393e3@63.111.7.161 >for seqno 102 (Request) > >Which is tied to the call in question. Any clues? > >--- Gavin > >_______________________________________________ >Asterisk-Users mailing list >Asterisk-Users@lists.digium.com >http://lists.digium.com/mailman/listinfo/asterisk-users
> From: John Todd [mailto:jtodd@loligo.com] > > exten => 4001,1,Dial(SIP/gadams,10,r)Hmm, the syntax below is what one of the Digium guys put in. Tried your format, getting a ring followed by the unavailable message. For now I have the 2 SIP phones in the ATL context to test phone to phone capabilities. I think it's correctly working as when I dial from one phone to the other voicemail is kicking in. I changed the context back to default, still some weirdness. The phones are on a private subnet, NATted (hidden or MASQed) behind a single IP address. Dialing from the phones is fine, but something has changed to where calls to the phone are failing. I assume the "maximum retries failed" message means * couldn't contact the phones. Ironically, the phones appear to be registered ok: *CLI> sip show peers Name/username Host Mask Port Status aojeda/aojeda 63.111.7.162 (D) 255.255.255.255 10945 Unmonitored gadams/gadams 63.111.7.162 (D) 255.255.255.255 10940 Unmonitored and the firewall is allowing the packets through. Anyway, I have some more technical support time with the Digium crew, I'll bring it up with them. Thanks for the response, --- Gavin
> From: Dave Weis [mailto:djweis@sjdjweis.com] > > On Mon, 4 Aug 2003, John Todd wrote: > > Try this: > > exten => 4001,1,Dial(SIP/gadams,10,r) > > I don't know how the syntax you've specified will behave; maybe it > > will work, but it's not any format I've used. Try the syntax above > > for your Dial line and see if it results in different actions. > > I can not get my 7960 to work with a non-numeric userid/username. Ihave> to define it in sip.conf as > [101] > instead of > [cisco1] > I change the authentication settings in the SIP menu on the phone andin> the sip.conf file but it will not register. The version is POS30202I'm using the 4.4 SIP firmware and appear to have good auth based on username instead of extension. Sip.conf: [gadams] type=friend username=gadams secret=blah context=default host=dynamic canreinvite=no nat=yes mailbox=4001 extension.conf: exten => 4001,1,Dial(SIP/gadams,10,r) and the specific SIPxxxxxxx.cnf file: ; Name - extension of phone in * terms line1_displayname : "Gavin Adams" line1_name : gadams line1_authname : gadams line1_password : xxxxxxxx Just struggling with a NAT/firewall issue now. It appears that since the src/dst pair for SIP is UDP 5060, and I have multiple phones hiding behind the same IP address, it's problematic. Going to attempt to do static NAT translation instead of "hide NAT" (FW-1). HTH, --- Gavin