No Audio After ISP and internal IP change [SOLVED]

[mbean]

First let's get the essentials out:

ViciBox Redux v.3.1.15
Linux viciexp 2.6.34.10-0.6-pae
Vicidial Version: 2.4-357a
Build: 120125-2107
Asterisk 1.4
Read manager manual: Yes

Now onto the problem.

We changed our ISP over the weekend. I updated the routes on our firewall to receive from the new IP address and forward to the the new internal IP address. Our carrier updated our IP information on their end.

I also ran the script to update the IP address in the database with the new IP address as well as changed the sip.cong externip setting to match.

Calls can connect and the consumer can hear the agent, but the agent can not hear the consumer.

I worked with the carrier a bit on this and have so far moved it from behind the router to the dmz and made the appropriate changes with no change in the issue. The same ports are open on the firewall as before when the system was working correctly.

Here is a snippet of the CLI output when running a test call:

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[Jan 27 11:45:27]     -- Executing [8504343118@trunkinbound:1] AGI("SIP/XCASTIN-00000005", "agi-DID_route.agi") in new stack
[Jan 27 11:45:27]     -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-DID_route.agi
[Jan 27 11:45:28]     -- AGI Script agi-DID_route.agi completed, returning 0
[Jan 27 11:45:28]     -- Executing [99909*11***DID@default:1] Answer("SIP/XCASTIN-00000005", "") in new stack
[Jan 27 11:45:28]     -- Executing [99909*11***DID@default:2] AGI("SIP/XCASTIN-00000005", "agi-VDAD_ALL_inbound.agi") in new stack
[Jan 27 11:45:28]     -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_ALL_inbound.agi
[Jan 27 11:45:28]     -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[Jan 27 11:45:28]     -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[Jan 27 11:45:28]     -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[Jan 27 11:45:28]     -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[Jan 27 11:45:29]     -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[Jan 27 11:45:29]     -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[Jan 27 11:45:29]     -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[Jan 27 11:45:29]     -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[Jan 27 11:45:29] ERROR[4645]: utils.c:967 ast_carefulwrite: write() returned error: Broken pipe
[Jan 27 11:45:29]     -- AGI Script agi-VDAD_ALL_inbound.agi completed, returning 0
[Jan 27 11:45:29]     -- Executing [s@CSIClosed:1] Answer("SIP/XCASTIN-00000005", "") in new stack
[Jan 27 11:45:29]     -- Executing [s@CSIClosed:2] AGI("SIP/XCASTIN-00000005", "agi-VDAD_inbound_calltime_check.agi|CALLMENU-----YES-----CSIClosed--------------------") in new stack
[Jan 27 11:45:29]     -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_inbound_calltime_check.agi
[Jan 27 11:45:29]     -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[Jan 27 11:45:29]     -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[Jan 27 11:45:29]     -- AGI Script agi-VDAD_inbound_calltime_check.agi completed, returning 0
[Jan 27 11:45:29]     -- Executing [s@CSIClosed:3] Set("SIP/XCASTIN-00000005", "INVCOUNT=0") in new stack
[Jan 27 11:45:29]     -- Executing [s@CSIClosed:4] BackGround("SIP/XCASTIN-00000005", "CSIClosed") in new stack
[Jan 27 11:45:29]     -- <SIP/XCASTIN-00000005> Playing 'CSIClosed' (language 'en')
[Jan 27 11:45:38] WARNING[2751]: chan_sip.c:2058 retrans_pkt: Maximum retries exceeded on transmission 7c4c9c1857ceae09f08a7d5b94a32294-19680435@38.102.250.162 for seqno 501 (Critical Response) -- See doc/sip-retransmit.txt.
[Jan 27 11:45:38] WARNING[2751]: chan_sip.c:2080 retrans_pkt: Hanging up call 7c4c9c1857ceae09f08a7d5b94a32294-19680435@38.102.250.162 - no reply to our critical packet (see doc/sip-retransmit.txt).
[Jan 27 11:45:38]   == Spawn extension (CSIClosed, s, 4) exited non-zero on 'SIP/XCASTIN-00000005'
[Jan 27 11:45:38]     -- Executing [h@CSIClosed:1] DeadAGI("SIP/XCASTIN-00000005", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----18---------------") in new stack
[Jan 27 11:45:38]     -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----18--------------- completed, returning 0


The things that obviously stick out are:

Code: Select all

[Jan 27 11:45:29] ERROR[4645]: utils.c:967 ast_carefulwrite: write() returned error: Broken pipe

and:

Code: Select all

[Jan 27 11:45:38] WARNING[2751]: chan_sip.c:2058 retrans_pkt: Maximum retries exceeded on transmission 7c4c9c1857ceae09f08a7d5b94a32294-19680435@38.102.250.162 for seqno 501 (Critical Response) -- See doc/sip-retransmit.txt.
[Jan 27 11:45:38] WARNING[2751]: chan_sip.c:2080 retrans_pkt: Hanging up call 7c4c9c1857ceae09f08a7d5b94a32294-19680435@38.102.250.162 - no reply to our critical packet (see doc/sip-retransmit.txt).


Any Ideas? All help is greatly appreciated.

Thanks,

Mike

[davesdatasystems]

[Jan 27 11:45:29] ERROR[4645]: utils.c:967 ast_carefulwrite: write() returned error: Broken pipe
means nothing;

if you are able post your sip.conf file
and your dial plan

i am guessing it is either in the sip.conf or the dial plan itself

[mbean]

Sure.

Dialplan Entry (Outbound):

Code: Select all

Domestic calls
exten => _91NXXNXXXXXX,1,AGI(agi://127.0.01:4577/call_log)
exten => _91NXXNXXXXXX,2,Dial(${XCAST}/${EXTEN:1},60,tTo)
exten => _91NXXNXXXXXX,3,Hangup
International calls
exten => _9011X.X,1,AGI(agi://127.0.01:4577/call_log)
exten => _9011X.X,2,Dial(${XCAST}/${EXTEN},60,tT)
exten => _9011X.X,3,Hangup


Dialplan Entry (Inbound):

Code: Select all

Domestic calls
exten => _91NXXNXXXXXX,1,AGI(agi://127.0.01:4577/call_log)
exten => _91NXXNXXXXXX,2,Dial(${XCASTIN}/${EXTEN:1},60,tTo)
exten => _91NXXNXXXXXX,3,Hangup
International calls
exten => _9011X.X,1,AGI(agi://127.0.01:4577/call_log)
exten => _9011X.X,2,Dial(${XCASTIN}/${EXTEN},60,tT)
exten => _9011X.X,3,Hangup


Tried to Post sip.conf but I am told my post looks to spammy.

[mbean]

Sip.conf

I removed any entries that were commented out from the beginning.

Code: Select all


[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)
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
disallow=all                    ; First disallow all codecs
allow=ulaw                      ; Allow codecs in order of preference
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
dtmfmode = rfc2833              ; Set default dtmfmode for sending DTMF. Default: rfc2833
videosupport=no                 ; Turn on support for SIP video. You need to turn this on
callevents=yes                  ; generate manager events when sip ua
rtptimeout=60                   ; Terminate call if 60 seconds of no RTP or RTCP activity
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
externip = x.x.x.x        ; Address that we're going to put in outbound SIP
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=yes                         ; Global NAT settings  (Affects all peers and users)
canreinvite=no          ; Asterisk by default tries to redirect the
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




[davesdatasystems]

why is it set 127.0.01:4577 that is not a true ip address it should be (agi://127.0.0.1:4577/call_log)

[mbean]

Honestly I don't know. When we originally set up the system we had issues with connecting to the carrier. I had actually called the good folks at vicidial to get their help and that was the dialplan entry i was told to use. It worked so I didn't question it though I probably should have. I am going to set it to 127.0.0.1:4577/call_log just to see but that diaplan entry was never changed in the transition.

[davesdatasystems]

also post the account entry part of your xcast dial pan, i seen something strange pop up that does not match a xcast ip addy, just making sure you had the right info from them

[davesdatasystems]

also private msg me your external ip address, i want to look to see if it matches up in the right place

[mbean]

I noticed it in the CLI trying to contact 38.102.250.162 which does not match the entries we had before or given today when trying a new IP to connect to. But the account entries are as follows:

Outbound:

**Code Removed**

Inbound:
**Code Removed**

[mbean]

Needed to make a small correction to the sip.conf file. Credit to davesdatasystems for solving this.

[williamconley]

Good job dave

We do not recommend "private IP addressing" for Vicidial, but it can be done. In this case it required a modification to sip.conf directly ... but sometimes it is possible to make this entry on the carrier in the Admin->Carriers->Account Entry section (to avoid making an entry directly in the sip.conf file, which will be overwritten during upgrade).

Fairly common problem, though ... we all bump into it eventually with private IP addressing.