[asterisk-users] Short format of SIP INVITE - how to change
My Asterisk box send INVITEs in the short form, i.e., f: instead of from, v: instead of via and so on. Is there a way to force asterisk to use full format? thanks Vitaly __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Why Asterisk doesn't accept sip302 redirect?
My asterisk should follow 302 redirect which it
receives from other sip server(10.10.10.10). By
running network sniffer I see, that asterisk receives
302 answer, but doesn't follow it.
My config is:
sip.conf:
...
[out4]
type=peer
host=10.10.10.10
canreinvite=no
promiscredir=yes
insecure=very
disallow=all
allow=g729
allow=g723
...
extensions.conf:
[to-sip]
exten = _0011X., 1, Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten = _0011X., 2, Hangup()
Any ideas?
Vitaly
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Re: [asterisk-users] Why Asterisk doesn't accept sip302 redirect?
Thanks for your answer, see details below:
U 10.10.10.10.67:5060 - 10.10.10.107:5060
INVITE sip:[EMAIL PROTECTED] SIP/2.0..v:
SIP/2.0/UDP
10.10.10.67:5060;branch=z9hG4bK0264a8da;rport..f:
2519494
sip:[EMAIL PROTECTED];tag=as1d5e5664..t:
sip:[EMAIL PROTECTED]..m:
sip:[EMAIL PROTECTED]..i: 503f1f3a
[EMAIL PROTECTED]: 102
INVITE..User-Agent: Asterisk PBX..Max-Forwards:
70..Date: Wed, 10 Oct 2
007 10:01:31 GMT..Allow: INVITE, ACK, CANCEL,
OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY..c:
application/sdp..l: 259v=0
..o=root 2423 2423 IN IP4
10.10.10.67..s=session..c=IN IP4 10.10.10.67..t=0
0..m=audio 17250 RTP/AVP 18 4 101..a=rtpmap
:18 G729/8000..a=fmtp:18 annexb=no..a=rtpmap:4
G723/8000..a=rtpmap:101
telephone-event/8000..a=fmtp:101 0-16..a=silence
Supp:off - - - -..
#
U 10.10.10.107:5060 - 10.10.10.67:5060
SIP/2.0 302 Redirect..Contact:
sip:[EMAIL PROTECTED]:11060..v: SIP/2.0/UDP
10.10.10.67:5060;branch=z9hG4bK0264
a8da;rport..CSeq: 102 INVITE..Content-Length: 0
Master.csv:
,2519494,001112345678,to-sip,2519494,SIP/10.10.10.66-09e0a8b0,SIP/out4-09e15578,Dial,SIP/12345678
@out4,2007-10-10 15:01:31,,2007-10-10
15:02:01,30,0,NO ANSWER,DOCUMENTATION
--- Alex Balashov [EMAIL PROTECTED] wrote:
Vitaly,
Can you provide details of what is going on in the
packet capture exactly?
What is the Contact: URI that the peer provides in
the 302 Moved response?
What does Asterisk do subsequently?
Cheers,
-- Alex
On Wed, 10 Oct 2007, Vitaly wrote:
My asterisk should follow 302 redirect which it
receives from other sip server(10.10.10.10). By
running network sniffer I see, that asterisk
receives
302 answer, but doesn't follow it.
My config is:
sip.conf:
...
[out4]
type=peer
host=10.10.10.10
canreinvite=no
promiscredir=yes
insecure=very
disallow=all
allow=g729
allow=g723
...
extensions.conf:
[to-sip]
exten = _0011X., 1, Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten = _0011X., 2, Hangup()
Any ideas?
Vitaly
Be a better Heartthrob. Get better relationship
answers from someone who knows. Yahoo! Answers -
Check it out.
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--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: +1-678-954-0670
Direct : +1-678-954-0671
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[asterisk-users] Strange problem with asterisk
Situation such. There is an asterisk working as office pbx. 6 fxo - 18 fxs ports. All works perfectly, but some times in a week something occurs. Could not catch what exactly yet. But symptoms such. The asterisk infinitely writes the message of a type to broad gullies: WARNING [20757] chan_zap.c: We're Zap/8-1, not ... ZOMBIE. Numbers of channels can change. Because of that that broad gullies get littered fairly promptly, I have not time to see that occured in an instant of the beginning of this event. When the asterisk is in such condition, the appropriating channel does not work, in this case 8. What can it be? asterisk version 1.2.14-BRIstuffed-0.3.0-PRE-1x ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] blindtransfer and initiator hangup
Good afternoon. The asterisk has two kinds transfer, attended and blind, me interests as to set for blindtransfer performance what or commands on exten = h for the one who this transfer initiated. I.e. now in the console it is visible Hangup the initiator but as on this Hangup to hang up performance of a command, for me a riddle. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RE:Asterisk and dialer Running on Thin Clients
You can, but it will demand a lot of work. We now work above introduction of such decision on thin clients under control of thinstation. As софтофона it is used mozphone (front-end), from the thin client network_client (back-end). ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RE:Asterisk and dialer Running on Thin Clients
Sorry You can, but it will demand a lot of work. We now work above introduction of such decision on thin clients under control of thinstation. As softphone it is used mozphone (front-end), from the thin client network_client (back-end). ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Habitual set of number
Good afternoon. For an output in city I use such construction:
exten = 9,1,Answer
exten = 9,2,SIPDtmfMode(rfc2833)
exten = 9,3, Set(TIMEOUT(digit)=3)
exten = 9,4,ChanIsAvail(ZAP/g2|j)
exten = 9,5,NoOp(${AVAILCHAN})
exten = 9,6,Playtones(dial)
exten = 9,7,Cut(chan=AVAILCHAN,-,1)
exten = 9,8,NoOp(${chan})
exten = 9,9,waitexten()
exten = _XX,1,Dial(${chan}/${EXTEN},,tT)
exten = _XX,2,Hangup
exten = _XXX,1,Dial(${chan}/${EXTEN},,tT)
exten = _XXX,2,Hangup
exten = 9,105,Playtones(busy)
exten = 9,106,Busy(10)
Like all it is quite good, except for one, hooter goes in a tube at
typing, at that time, while it hammers in number in waitexten. How it
is possible to realize too most, only that with a set of the first
figure hooter interrupted? In advance thanks.
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[asterisk-users] call-limit and problem with freezy phones. also freezy zap channels with x101p card.
call-limit and problem with freezy phones. also freezy zap channels with x101p card. Hello all. I have installed asterisk 1.2.9.1 and zaptel 1.2.6. I have such configuration: I have some phones with planet vip-156 with configuration in sip.conf: [036] ; planet 222 type=friend host=dynamic canreinvite=yes username=036 secret=036 nat=no qualify=10 dtfmode=rfc2833 musiconhold=default context=office callerid=036 disallow=all allow=ulaw callgroup=1 pickupgroup=1 call-limit=1 . everything work good, but sometimes i have situation in which the asterisk thinks that phone is borrowed at present, and appear message like that: cannot create a sip channel due to usage limit... but when in this situation i check channels with comand show channels, i see that phone, on which can not call, in this moment absolutly free. That situation appear when i started using call-limit=1. When i do asterisk -rx reload, then that fixes. How that can be fixed without reloads? Also I have problem with zap channels only with x101p cards. Sometimes channel stay up even when line hangup. Also I have tdm400 cards, they work perfect. section in zappata.conf for x101p channel: context=generic-inc signalling=fxs_ks rxwink=300 ; Atlas seems to use long (250ms) winks usedistinctiveringdetection=yes hidecallerid=no callwaiting=no usecallingpres=no callwaitingcallerid=no threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=no echocancelwhenbridged=no echotraining=no rxgain=-4 txgain=-4 ;group=2 callgroup=3 pickupgroup=3 immediate=yes busydetect=yes busycount=8 callprogress=no pulsedial=no musiconhold=default switchtype = national group = 3 channel = 6 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] zaptel Disabled echo canceller because of tone (rx) on channel 2 work?
All greetings. If I have correctly understood, echocancel algoritms prevent to go to fax-messages. Therefore at detection of attempt of reception or transfer of a fax-message, echocancel on a line it is disabled, in /var/log/messages thus appears such messages zaptel Disabled echo canceller because of tone (rx) on channel 2. At me a problem that detection of a fax or some reason works very seldom as consequence faxes go very badly. For a week of work, echocancel disables 2-3 times, thus every day is accepted and sends tens fax-messages. As experiment I included in/etc/zapata.conf faxdetect=yes on a line. At sending or reception of a fax, the line understood it very quickly. How it is possible to improve reaction of a line to faxes that echocancel disabled always? Sorry for my bad english and thanks for help. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Softphone on Thinclient?
I currently use NTAVO thin clients w/ Thinstation and I would love to put a soft phone on them, but I don't think that would work well (they use RDP), or do you all know if there is a smooth way to make the interface work? I don't really picture my users switching between an RDP session X-Windows (i.e. ALT-F3/ALT-F4) I have compilled for Thinstation softphone named KIAX. Switch beetwen RDP session and softphone doing like ALT-F3/ALT-F4. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 191st simultaneous call fails
Hello, Have you analized quality of the calls ? what was quality of 190 call ? :) On Thu, 16 Dec 2004 19:51:28 -0800, Jim Gottlieb [EMAIL PROTECTED] wrote: I've been testing both T400P and TE405P boards and I'm running into some kind of hard limit on the number of simultaneous calls. This is on x86 with 2 Athlon MP 2800+ CPUs running Fedora Core 1. Everything is fine up to 190 channels, but the 191st call fails every time with errors like: Dec 14 15:44:00 WARNING[1215]: Unable to start PBX on Zap/201-1 Dec 14 15:44:00 WARNING[1215]: Failed to create update thread! Dec 14 15:44:00 WARNING[1215]: Unable to start PBX on channel 0/9, span 9 Dec 14 15:44:00 WARNING[1215]: Call specified, but not found? Dec 14 15:44:00 WARNING[1215]: Hangup on bad channel 0/9 on span 9 It's not tied to which channel the call comes in on. It's some resource that's exhausted after 190 calls. A limit on threads? I thought it might be per-process file descriptors even though we were only going up to 529 on that PID and I used 'ulimit -n' to increase it before starting asterisk, but that didn't make a difference. # cat /proc/sys/kernel/threads-max 14336 I would think that's enough, but perhaps the per-process limit is much lower. Any clues? Thanks... ___ 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 -- Vitaly Nikolaev ___ 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
