Audio drops at 20 seconds exactly

Located in /etc/asterisk there is a file called sip_general_additional.conf

There you will see something like this

vmexten=*97
bindport=5060
bindaddr=0.0.0.0
disallow=all
allow=ulaw
allow=alaw
context=from-sip-external
callerid=Unknown
notifyringing=yes
notifyhold=yes
limitonpeers=yes
tos_sip=cs3
tos_audio=ef
tos_video=af41

In this config the only allowed voice codecs are the alaw and ulaw.

But to really help you out, there is not enough information to assist.

Are these calls failing from phone to phone?
What is the prefered codec order on the phone?
What is your upstream? are you using SIP, IAX2 or do you have a card for pstn?
Did you set in the asterisk cli core set verbose 9 and sip debug? to see if there is a cancel or bye in the sip messages?

I'm sure that after we can see what is happening, someone will be able to help, just as i hope that this helps.

I'm betting on NAT problems.

------
[holex.dyndns.org ~]# asterisk -rvvvvvvvvvv
Asterisk 1.4.20-1 RPM by vc-rpms@voipconsulting.nl, Copyright (C) 1999 - 2008 Digium, Inc. and others.
Created by Mark Spencer
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
== Parsing '/etc/asterisk/asterisk.conf':Found
Connected to Asterisk 1.4.20-1 RPM by vc-rpms@voipconsulting.nl currently running on holex (pid = 2432)
Verbosity is at least 12
-- Executing [s@from-sip-external:1] GotoIf("SIP/204.11.194.35-088d4d20", "1?from-trunk||1") in new stack
-- Goto (from-trunk,s,1)
-- Executing [s@from-trunk:1] Set("SIP/204.11.194.35-088d4d20", "__FROM_DID=s") in new stack
-- Executing [s@from-trunk:2] Gosub("SIP/204.11.194.35-088d4d20", "app-blacklist-check|s|1") in new stack
-- Executing [s@app-blacklist-check:1] LookupBlacklist("SIP/204.11.194.35-088d4d20", "") in new stack
-- Executing [s@app-blacklist-check:2] GotoIf("SIP/204.11.194.35-088d4d20", "0?blacklisted") in new stack
-- Executing [s@app-blacklist-check:3] Return("SIP/204.11.194.35-088d4d20", "") in new stack
-- Executing [s@from-trunk:3] GotoIf("SIP/204.11.194.35-088d4d20", "1 ?cidok") in new stack
-- Goto (from-trunk,s,5)
-- Executing [s@from-trunk:5] NoOp("SIP/204.11.194.35-088d4d20", "CallerID is "2222488492" ") in new stack
-- Executing [s@from-trunk:6] Set("SIP/204.11.194.35-088d4d20", "_RGPREFIX=Holex") in new stack
-- Executing [s@from-trunk:7] Set("SIP/204.11.194.35-088d4d20", "CALLERID(name)=Holex2222488492") in new stack
-- Executing [s@from-trunk:8] Goto("SIP/204.11.194.35-088d4d20", "ivr-4|s|1") in new stack
-- Goto (ivr-4,s,1)
-- Executing [s@ivr-4:1] Set("SIP/204.11.194.35-088d4d20", "LOOPCOUNT=0") in new stack
-- Executing [s@ivr-4:2] Set("SIP/204.11.194.35-088d4d20", "__DIR-CONTEXT=default") in new stack
-- Executing [s@ivr-4:3] Set("SIP/204.11.194.35-088d4d20", "_IVR_CONTEXT_ivr-4=") in new stack
-- Executing [s@ivr-4:4] Set("SIP/204.11.194.35-088d4d20", "_IVR_CONTEXT=ivr-4") in new stack
-- Executing [s@ivr-4:5] GotoIf("SIP/204.11.194.35-088d4d20", "0?begin") in new stack
-- Executing [s@ivr-4:6] Answer("SIP/204.11.194.35-088d4d20", "") in new stack
-- Executing [s@ivr-4:7] Wait("SIP/204.11.194.35-088d4d20", "1") in new stack
-- Executing [s@ivr-4:8] Set("SIP/204.11.194.35-088d4d20", "TIMEOUT(digit)=3") in new stack
-- Digit timeout set to 3
-- Executing [s@ivr-4:9] Set("SIP/204.11.194.35-088d4d20", "TIMEOUT(response)=60") in new stack
-- Response timeout set to 60
-- Executing [s@ivr-4:10] BackGround("SIP/204.11.194.35-088d4d20", "custom/PCM_Fin_Raiz1") in new stack
-- Playing 'custom/PCM_Fin_Raiz1' (language 'en')
== Spawn extension (ivr-4, s, 10) exited non-zero on 'SIP/204.11.194.35-088d4d20'
-- Executing [h@ivr-4:1] Hangup("SIP/204.11.194.35-088d4d20", "") in new stack
== Spawn extension (ivr-4, h, 1) exited non-zero on 'SIP/204.11.194.35-088d4d20'
holex*CLI>
--------

To add something new; this extrange drops and no dial tones does NOT happen between sip peers, nor with the ZAP channels. It only happens with this DID demo.

VOIP Newbie

At this point I am atonished! This file was not empty; weel now is completely blank! I am looking for a sip_nat.conf.bak but nothing!

I remember I modified days ago this file to setup the dyndns domain; another thing is in modules DHCP is not installed?

What could this be?

VOIP Newbie

Hello,

Now even inside the NAT (LAN) sip peers fail to get more than 20 secs audio, no callee audio at all; I don´t now how to put attachments here, so I´ve uploaded the entire SIP files in a recap, you can get it here http://holidayexpress.com.mx/sip.pdf

I am *very frustrated* because this was not happening; I am not sure what could had happened... We got installed a hell-zapmicro card, the card is working fine; we can now get and make calls tru the ZAP channels ok, but sip is no longer working... seems to me pretty odd ZAP install could mess SIP?

VOIP Newbie

Hi stechnique, thanks for your prompt response,

Here´s the RTP.conf:
-------------
;
; RTP Configuration
;
[general]
;
; RTP start and RTP end configure start and end addresses
;
rtpstart=10001
rtpend=20000
----------------
Please note the firewall settings on the router:
TCP 22
TCP 6600
TCP 10000
UDP 4569
UDP 5004-5082
UDP 10001-20000

Also, the trixbox network settings are

[holex.dyndns.org ~]# ifconfig eth0
eth0 Link encap:Ethernet HWaddr 00:0E:2E:23:0D:A6
inet addr:192.168.1.10 Bcast:192.168.1.255 Mask:255.255.255.0
inet6 addr: fe80::20e:2eff:fe23:da6/64 Scope:Link
UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1
RX packets:2328 errors:0 dropped:0 overruns:0 frame:0
TX packets:5873 errors:0 dropped:0 overruns:0 carrier:0
collisions:0 txqueuelen:1000
RX bytes:430804 (420.7 KiB) TX bytes:1425388 (1.3 MiB)
Interrupt:10 Base address:0x8000
*Note I use an out of DHCP address, I set the Trixbox Network settings eth0 to fixed IP, then the router accordingly reflects Trixbox_Srv -> 192.168.1.10 as fixed.

VOIP Newbie

Just a wild guess here, but it could be your isp. Time Warner in NYC tried this crap with me (of course they denied it though)

We have a first more info file at http://holidayexpress.com.mx/sip.pdf regarding SIP config files
And now we have a firewall report, PBX status with odd behaviors here

http://holidayexpress.com.mx/mess.pdf And that mess name is just that, this is a mess; packets are going thru and no voice no nada amigos!

VOIP Newbie

As to where are those fwall ports you saw open, those are forwarded to the Trixbox, I just covered the router public IP, at left, you can see Trixbox_Srv, that is the PBX, It is set to receive traffic for those ports. As of the nat=no, I´ve removed that line and stated no in extentions config, and if you take a closer look at the firewall log, where I circled the phone IP, the Box IP and the fwall says the packets are going thru, but nada, nothing..
What else could it be?

VOIP Newbie

Did you get this resolved or have you stuck with using your externip instead of externhost ?

If its not solved try using NO-IP or similar instead of DYNDNS.

Ive spent all day looking at this 20 second drop problem which came on yesterday for no apparent reason. I just changed to a NO-IP address and it seems to have solved it. It makes no sense to me but I'm fairly sure that was the problem, I'll test further in the morning.

No joy; for me, no-ip or dyndns makes no difference; the problem is not there both dns records point to my wan ip.

Again, to make clear the problem; if I do use externip=mywanip it does work (only inside the NATed LAN) outside SIP peers are able to register but not able to comunicate; sometimes I get unidirectional audio, sometimes I get nothing, nada!

Almost bold by now...

VOIP Newbie

http://www.dyndns.com/support/tools/openport.html

There should be no difference between a no-ip or dyndns. Use the command line tool dig to check if the entry is reporting the correct wan address, or if there are any duplicate answers being reported. But apparently thats working if you are registering.

Sounds like ports 5060 is open and working correctly as things are registering. Check /etc/asterisk/rtp.conf and look at your RTP (audio) port range. You can change these if for some reason they are being blocked.

Another thought is do you you have any other applications on your network that might be using upnp with conflicting ports in the RTP range? If no, and everything is open and forwarded properly, I'd say you have a possible router CPU load issue.

I looked at your X-Lite config, looked good to me, even thought I don't mess with the Authorization user name. Honestly I've never been very impressed with the 2-Wire DSL modems. If you can put that in a transparent bridging mode that would take the 2Wire's firewall and NAT translation out of the picture for troubleshooting.

Hi guys, thanks for your replying...

There is nothing attached but a laptop and another PC, thru eth, cabled; have a question; If the SIP phones can register means 5060/5061 are open? I am still confused; actually, my ISP denies closed ports except for UDP 5073/5077 wich they use to control stuff, but I have ran so many different utilities to check and some state ports are actually closed, some others state they are ok; what I suspect is that ISP might be scrutinizing this ports so hard that mess something on em; another thing my ISP cust service said is that they recomend to use UDP's between 8000-8750 for rtp; changed that (rtp.conf) and nothing, the same thing.

Another question is, if I setup the modem router (wich is such a piece of crap) as a bridge, and do DMZ (this is, turn off routing and firewall) and then attach a, let's say a Linksys router to it, would not this be to double up the prob? (double NAT traversing?) Is there actually a specific Linksys DSL modem router model wich you could recommend to me to *replace* the 2wire junk?

Also I would not like to go ahead and purchase more hardware if I cant make sure is not hte ISP and not the router; in that case, I am considering changing my ISP for a simmetric DSL (Telmex provides ADSL) but this change is only available wireless (I´ve read so many delay issues with wireless DSL) and is so expensive that I am at a dead alley here.

There stills the Skyking recommendation to go VPN but, the same again, have read lots (many) posts regarding Telmex DSL routers resetting and or blowing or dropping connections because of overhead due to VPN usage for transporting voice in a rather non industrial setting such as ADSL.

Also, ´ve been reading about third parties offering services within ranges from 4 to 50 usd a month to do the job 8working around ISP blockages, port issues and so) but I am very unsure... I have less than no experience with Unix / Linux / Asterisk that everything is so new to me I dont know how to work around this.

Thanks for your comments

VOIP Newbie

As I had indicated the only definitive way to know if a UDP port is open is to run Wireshark or Ethereal on your laptop and watch the datagrams come in from the Internet.

Second issue, if you bridge that is at layer 2 so you would not be double NAT'ing.

NAT is a Layer 3 function.

Right now I am following Scott recommendation; I am following closely what wireshark shows up - by the way, It´s a wonderful tool. So far, I see everything ok BUT I see packages saying 401 auth denied for UTP´s within the range, so, so far, 5060 and 5061 works just fine, the issue is in the RTP UDP's. There is the flaw.

The action plan, is as follows:
1. Will do a DMZ direct to the Trixbox and if this fails, then
2. Will go for a different older 2wire modem-router-fwall that I´ve read is actually working with a setup like mine, and if this fails, then
3. I am going to look for purchase either a Netgear FVX538 or a Linksys RV082 Router, and if that fails, then
4. I am going to change ISP for another one more VoIP friendly and if that fails, then
5. I am going to put myself on fire and throw me from the 15th floor while burning.

Meanwhile, I am going to take your offer, I´ll contact you tomorrow.

Thanks again Percy

VOIP Newbie

Well Scott, yes you can see the wireshark packet 40 detail (showed many more just like that one) says as follows:
_________________________
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 189.130.133.143:8000;branch=z9hG4bK-d8754z-a6477a1b874b8012-1---d8754z-;received=189.130.133.143;From: "Carlos Sierra";tag=9a2b4879
To: "Carlos Sierra";tag=as706368c6
Call-ID: ZGIzMDkyMmQzNWFiZjM2N2Y2MTlkMzVjOThkOTg4NDQ.
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5724333a"
Content-Length:0
_________________________

You can get the entire detail at http://holidayexpress.com.mx/p40.pdf

On the other hand, why is that I can have full bidirectional inside the LAN, but on remote SIP peer i still have no more than 20 secs audio and no DTMF's response... I guess I have both problems; a codec issue and a NAT issue.

And yes, Percy you are very wellcome to help me too!

Thank you very much for your help.

VOIP Newbie