[asterisk-users] Calls drop after a couple of minutes.
I have been encountering a rather hard to debug problem for the last couple of months: * Calls are setup fine. * After a couple of minutes, two way audio becomes one-way and the remote or local party drops out of the call. Setup: * Nokia E71i sip on NAT'd network (multihomed linux box) * Remote asterisk 1.4.21 on Ubuntu on public network * using a Finera/Betamax provider to route calls to PSTN. I initially thought it may be a NAT problem and have checked everything on the NAT gateway/firewall. I see no rejected packets hitting the firewall logs. I'm really at a loss as to what could be causing the calls to drop out for one party so regularly. Any clues where I could look further to debug this would be most useful. local firewall: modprobe ip_conntrack_sip ports=5060 modprobe ip_nat_sip # probably not needed since everything is forwarded: $IPTABLES -A FORWARD -s $INTERNAL_NET -d $ANYWHERE -p udp --dport 5060 -j accept-log # sip remote Asterisk server: $MODPROBE ip_conntrack $MODPROBE ip_conntrack_sip ports=5060 $IPTABLES -A INPUT -s $ANYWHERE -d $PUBLIC_ADDR -p udp --dport 5060 -j accept-log # voip $IPTABLES -A OUTPUT -s $PUBLIC_ADDR -d $ANYWHERE -p udp --sport 5060 -j accept-log # voip $IPTABLES -A OUTPUT -s $PUBLIC_ADDR -d $ANYWHERE -p udp --dport 5060 -j accept-log # voip $IPTABLES -A INPUT -s $ANYWHERE -d $PUBLIC_ADDR -p udp --dport 1:2 -j accept-log # voip $IPTABLES -A OUTPUT -s $PUBLIC_ADDR -d $ANYWHERE -p udp --sport 1:2 -j accept-log # voip sip.conf: [101] callerid=Simon Tennant type=friend username=101 secret=xx host=dynamic reinvite=no canreinvite=no mailbox=101 context=from-internal nat=yes port=5060 qualify=yes insecure=very disallow=all allow=alaw also sip.conf [justvoip.com] type=peer host=sip.justvoip.com fromdomain=sip.justvoip.com progressinband=yes disallow=all allow=alaw ; only alaw works with sip1... nat=no canreinvite=no qualify=yes insecure=port,invite username=imagi-justvoip fromuser=00491785450880 secret= registerattempts=0 ; keep trying to register (normally times out after 10 attempts) context=from-external from rtp.conf rtpstart=19000 rtpend=2 -- Simon Tennant _ fixed: .uk +44 20 7043 6756 .de +49 89 420 955 854 mob: .uk +44 78 5335 6047 .de +49 17 8545 0880 xmpp: [EMAIL PROTECTED] ___ -- 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] Asterisk on UML (User Mode Linux)
What's the current thinking on running Asterisk in a UML environment? I saw some discussion about Xen and asterisk on a Xen DomU. I'm currently running Asterisk in a UML and have noticed poorer quality on calls. I'm only using SIP and IAX2 trunks. No hardware adapters. I guess timing is important, but even if I could get the provider to install a kernel with the Zaptel Dummy timing device compiled in (impossible to install kernel modules in UML), I'm not convinced this would necessarily provide an accurate enough timing device. Is anyone else running their Asterisk instance in UML? If anyone is, what's the preferred way to keep timing accurate? Thinking I may have been too hasty in switching to UML... S. -- Simon Tennant ___ http://imaginator.com/~simon/contact signature.asc Description: OpenPGP digital signature ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] 3 way calling independent of phone hw.
I'm looking for a recipe for a 3 way call where one of the parties can (without using the flash button) dial-out and add a third participant to the call. I tried Googling but it seems I'm missing a key search term. The reason I wanted to avoid using the flash button is that some handsets don't have it (nokia E61 who's 2 way calling via sip is also broken) Something like: 1. party 1 calls party 2 2. either party 1 or 2 hits * on keypad 3. asterisk prompts for party 3's telephone number 4. asterisk dials party 3. 5. party 3 answers and is immediately added to 3-way call 6. the inviter has the option of pushing # to terminate party 3 (should the call only reach party 3's voicemail). Either that or a ways to do DISA from within the meet-me functionality. I can't imagine I'm the only person with this sort of requirement. -- Simon Tennant http://imaginator.com/~simon/contact signature.asc Description: OpenPGP digital signature ___ --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] Asterisk not hanging up calls
I have noticed that Asterisk (version 1.2.13) is not hanging up a call when the wifi handset moves out of range. My setup is Nokia E61 connected to wifi access point (private IP range) and then to server on internet (public IP). I have been testing using the talking clock application, and walking out of range does not hang up the call. The call will continue for hours even though the handset re-registers on another access point 5 minutes later with a different public IP. Subsequent calls continue fine although I still see traffic heading out to the old public IP address of the wifi access point. I thought the SIP control channel would do some kind of keeping state and time out a call after x number of failed replies. I know there is a timeout option but would rather not set a timeout on all calls. Here's the sip.conf that I am using for the device: * Name : 105 Secret : Set MD5Secret: Not set Context : from-internal Subscr.Cont. : Not set Language : AMA flags: Unknown CallingPres : Presentation Allowed, Not Screened Callgroup: Pickupgroup : Mailbox : 101 VM Extension : asterisk LastMsgsSent : 2048 Call limit : 0 Dynamic : Yes Callerid : Simon Tennant (Nokia E61) Expire : 3595 Insecure : no Nat : Route ACL : No CanReinvite : No PromiscRedir : No User=Phone : No Trust RPID : No Send RPID: No DTMFmode : auto LastMsg : 0 ToHost : Addr-IP : 10.15.11.8 Port 5060 Defaddr-IP : 0.0.0.0 Port 5060 Def. Username: 105 SIP Options : (none) Codecs : 0x70f (g723|gsm|ulaw|alaw|g729|speex|ilbc) Codec Order : (alaw,ulaw,ilbc,speex,gsm,g729,g723) Status : OK (266 ms) Useragent: Reg. Contact : sip:[EMAIL PROTECTED] S. -- Simon Tennant http://imaginator.com/~simon/contact signature.asc Description: OpenPGP digital signature ___ --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] WaitExten only reading 1 digit.
I am trying to setup an interactive menu where a caller hits the main
menu and can then dial an extension. As far as I can tell the
Waitexten app is failing to read 3 digits and just reading the first
and then announcing that it is invalid since all extensions are 3 digits.
How do I make Waitexten wait for 3 digits?
I have setup the extension 100 for users to reach the switchboard as
they would from outside:
[internal-extensions]
exten = 100,1,Goto(mainmenu,s,10)
exten = 101,1,Dial(SIP/101,30)
exten = 101,2,Voicemail(u101)
exten = 101,3,Hangup()
exten = 102,1,Dial(SIP/102,30)
exten = 102,2,Voicemail(u102)
exten = 102,3,Hangup()
dialing 100 then hits mainmenu
[mainmenu]
exten = s,10,Answer
exten = s,11,Wait(1)
exten = s,12,Background(buddy-cloud/welcome2)
exten = s,13,WaitExten(15)
exten = s,14,NoOp(Number dialed ${EXTEN})
include = internal-extensions
exten = t,1,Playback(vm-goodbye)
exten = t,2,Hangup
exten = i,1,Playback(invalid) ; That's not valid, try again
This is the output from me (x101) dialing the switchboard (x100)
-- Executing Goto(SIP/101-08186e70, mainmenu|s|10) in new stack
-- Goto (mainmenu,s,10)
-- Executing Answer(SIP/101-08186e70, ) in new stack
-- Executing Wait(SIP/101-08186e70, 1) in new stack
-- Executing BackGround(SIP/101-08186e70, buddy-cloud/welcome2)
in new stack
-- Playing 'buddy-cloud/welcome2' (language 'en')
-- Sent into invalid extension 's' in context 'mainmenu' on
SIP/101-08186e70
-- Executing Playback(SIP/101-08186e70, invalid) in new stack
-- Playing 'invalid' (language 'en')
-- Timeout on SIP/101-08186e70
== CDR updated on SIP/101-08186e70
-- Executing Playback(SIP/101-08186e70, vm-goodbye) in new stack
-- Playing 'vm-goodbye' (language 'en')
-- Executing Hangup(SIP/101-08186e70, ) in new stack
== Spawn extension (mainmenu, t, 2) exited non-zero on 'SIP/101-08186e70'
Where am I going wrong and do I need to worry about Sent into invalid
extension 's' in context 'mainmenu' on SIP/101-08186e70 warnings?
S.
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Simon Tennant http://imaginator.com/~simon/contact
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Re: [asterisk-users] WaitExten only reading 1 digit.
Doug Lytle wrote: Doug Lytle wrote: Simon Tennant wrote: [internal-extensions] exten = 100,1,Goto(mainmenu,s,10) You can't start at 10 on your menu, you have to start with 1. strange - I jumped into that context at 10 and numbered up from 10 - I thought that was ok. Also when I started numbering from 1 everything works. Cheers. -- Simon Tennant http://imaginator.com/~simon/contact signature.asc Description: OpenPGP digital signature ___ --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
