vicidial.org

Hey guys, so I got something funky going on can't quite figure out what is happening. At some point after I upgraded one of my servers to ViciBox 7, with all the new fancy asterisk 11 stuff inbound calls from one of our carriers is having audio problems. This carrier is setup for our office phones, so within vicidial the DID is routed to an extension. This setup had no issues prior to the upgrade. I reached out to our carrier on this and they say the failed calls are returning a private ip for the media, that IP is the IP of the phone it is ringing. Now every handful of calls we get no audio. The carrier round robins our servers, and this only appears to happen when the call comes in the server these extensions are setup on. If it comes in to a different server the audio is fine. I've toyed with the nat settings in sip.conf, tried both the nat=force_rport,comedia and nat=yes even though 11 says it is deprecated. (I noticed there was some bug related to nat settings that was fixed in asterisk 11.5 not sure if this patch is included in the vicibox version), but for everything else the settings are exactly the same as they were before the upgrade when everything was working fine. We also don't appear to be having this issue with DIDs from other carriers that are routed through ingroups, but that is probably due to it using conferences and not routing to actual extensions. Below is my sip.conf on that server, as well as the carrier setup in question. If there is any additional info I can provide please let me know.

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[general]
context=trunkinbound            ; Default context for incoming calls
allowguest=no                  ; Allow or reject guest calls (default is yes)
allowoverlap=no                 ; Disable overlap dialing support. (Default is yes)
;allowtransfer=no               ; Disable all transfers (unless enabled in peers or users)
;realm=mydomain.tld             ; Realm for digest authentication
bindport=5060                   ; UDP Port to bind to (SIP standard port is 5060)
bindaddr=0.0.0.0                ; IP address to bind to (0.0.0.0 binds to all)
srvlookup=yes                   ; Enable DNS SRV lookups on outbound calls
;domain=mydomain.tld            ; Set default domain for this host
;pedantic=yes                   ; Enable checking of tags in headers,
;tos_sip=cs3                    ; Sets TOS for SIP packets.
;tos_audio=ef                   ; Sets TOS for RTP audio packets.
;tos_video=af41                 ; Sets TOS for RTP video packets.
;maxexpiry=3600                 ; Maximum allowed time of incoming registrations
;minexpiry=60                   ; Minimum length of registrations/subscriptions (default 60)
;defaultexpiry=120              ; Default length of incoming/outgoing registration
;t1min=100                      ; Minimum roundtrip time for messages to monitored hosts
;notifymimetype=text/plain      ; Allow overriding of mime type in MWI NOTIFY
;checkmwi=10                    ; Default time between mailbox checks for peers
;buggymwi=no                    ; Cisco SIP firmware doesn't support the MWI RFC
;vmexten=voicemail              ; dialplan extension to reach mailbox sets the
disallow=all                    ; First disallow all codecs
allow=ulaw                      ; Allow codecs in order of preference
allow=h264
allow=h263
allow=gsm
mohinterpret=default
mohsuggest=default
language=en                     ; Default language setting for all users/peers
relaxdtmf=yes                   ; Relax dtmf handling
trustrpid = no                  ; If Remote-Party-ID should be trusted
sendrpid = yes                  ; If Remote-Party-ID should be sent
progressinband=no               ; If we should generate in-band ringing always
;useragent=Asterisk PBX         ; Allows you to change the user agent string
;promiscredir = no              ; If yes, allows 302 or REDIR to non-local SIP address
;usereqphone = no               ; If yes, ";user=phone" is added to uri that contains
dtmfmode = rfc2833              ; Set default dtmfmode for sending DTMF. Default: rfc2833
;compactheaders = yes           ; send compact sip headers.
videosupport=yes                 ; Turn on support for SIP video. You need to turn this on
;maxcallbitrate=384             ; Maximum bitrate for video calls (default 384 kb/s)
callevents=yes                  ; generate manager events when sip ua
;alwaysauthreject = yes         ; When an incoming INVITE or REGISTER is to be rejected,
;g726nonstandard = yes          ; If the peer negotiates G726-32 audio, use AAL2 packing
;matchexterniplocally = yes     ; Only substitute the externip or externhost setting if it matches
;regcontext=sipregistrations
rtptimeout=60                   ; Terminate call if 60 seconds of no RTP or RTCP activity
;rtpholdtimeout=300             ; Terminate call if 300 seconds of no RTP or RTCP activity
;rtpkeepalive=<secs>            ; Send keepalives in the RTP stream to keep NAT open
;sipdebug = yes                 ; Turn on SIP debugging by default, from
;recordhistory=yes              ; Record SIP history by default
;dumphistory=yes                ; Dump SIP history at end of SIP dialogue
;allowsubscribe=no              ; Disable support for subscriptions. (Default is yes)
;subscribecontext = default     ; Set a specific context for SUBSCRIBE requests
notifyringing = yes             ; Notify subscriptions on RINGING state (default: no)
notifyhold = yes                ; Notify subscriptions on HOLD state (default: no)
limitonpeers = yes              ; Apply call limits on peers only. This will improve
;t38pt_udptl = yes            ; Default false
;register => 1234:password@mysipprovider.com
;registertimeout=20             ; retry registration calls every 20 seconds (default)
;registerattempts=10            ; Number of registration attempts before we give up
externip = 66.119.xxx.xxx        ; Address that we're going to put in outbound SIP
;externhost=test.test.com     ; Alternatively you can specify a domain
;externrefresh=10               ; How often to refresh externhost if
localnet=192.168.0.0/255.255.0.0; All RFC 1918 addresses are local networks
localnet=10.0.0.0/255.0.0.0     ; Also RFC1918
localnet=172.16.0.0/12          ; Another RFC1918 with CIDR notation
localnet=169.254.0.0/255.255.0.0 ;Zero conf local network
nat=force_rport,comedia         ; Global NAT settings  (Affects all peers and users)
canreinvite=yes          ; Asterisk by default tries to redirect the
;directrtpsetup=yes             ; Enable the new experimental direct RTP setup. This sets up
;rtcachefriends=yes             ; Cache realtime friends by adding them to the internal list
;rtsavesysname=yes              ; Save systemname in realtime database at registration
;rtupdate=yes                   ; Send registry updates to database using realtime? (yes|no)
;rtautoclear=yes                ; Auto-Expire friends created on the fly on the same schedule
;ignoreregexpire=yes            ; Enabling this setting has two functions:
;domain=mydomain.tld,mydomain-incoming
;domain=1.2.3.4                 ; Add IP address as local domain
;allowexternaldomains=no        ; Disable INVITE and REFER to non-local domains
;autodomain=yes                 ; Turn this on to have Asterisk add local host
;fromdomain=mydomain.tld        ; When making outbound SIP INVITEs to
jbenable = yes              ; Enables the use of a jitterbuffer on the receiving side of a
jbforce = no                ; Forces the use of a jitterbuffer on the receive side of a SIP
jbmaxsize = 100             ; Max length of the jitterbuffer in milliseconds.
jbresyncthreshold = 1000    ; Jump in the frame timestamps over which the jitterbuffer is
jbimpl = fixed              ; Jitterbuffer implementation, used on the receiving side of a SIP
jblog = no                  ; Enables jitterbuffer frame logging. Defaults to "no".
qualify=yes             ; By default, qualify all peers at 2000ms
limitonpeer = yes       ; enable call limit on a per peer basis, different from limitonpeers

#include sip-vicidial.conf

; register SIP account on remote machine if using SIP trunks
; register => testSIPtrunk:test@10.10.10.16:5060
;
; setup account for SIP trunking:
; [SIPtrunk]
; disallow=all
; allow=ulaw
; allow=alaw
; type=friend
; username=testSIPtrunk
; secret=test
; host=10.10.10.16
; dtmfmode=inband
; qualify=1000


Carrier Account Entry: (setup by ip on their end no registration)

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[TouchToneNew1]
disallow=all
allow=ulaw
type=friend
host=192.104.111.33
dtmfmode=rfc2833
insecure=invite,port

Carrier Dialplan:

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exten => _X.,1,AGI(agi-DID_route.agi)

Any help you could provide would be greatly appreciated. Thanks!

I'm a goof. I think it was canreinvite=yes not sure why I switched it on, but that is the only difference between the config before reinstall and now. Switched it to no and haven't had the issue yet. But due to the sporradic nature of it, I'm still not 100% sure it is fixed. I guess I'll leave it for the next few business days and see if it crops up at all.

Hello

Did you solve this problem? I have the same problem - about 10 calls a day out of 5000 inbound.

Thanks

Roger

Often this is the result of the call coming from a different IP address through the carrier. Note that control (port 5060) and audio (random port between 10000-25000) signals may come from different locations. Thus your firewall may be blocking the audio (from an unkown IP) but allowing the control (from a known IP).