[Asterisk-Users] Connecting 2 Asterisk using SIP

Tue Dec 6 21:09:00 MST 2005

Well,

At first, I was just learning....

Then, it became a troubleshooting process.

I have box1 acting as a PSTN gateway with a TE405P. Box2 is used for  
our call center agents to login,answer,make calls and where calls are  
all recorded.

I've been trying to troubleshoot a call-drop problem I continue  
having and have had for months. It is becoming seriously irritating  
and demoralizing to the point that upper management is almost decided  
to drop the whole asterisk-based project.

As a test, I divided my agents in two groups. One group will take/ 
send calls to box1 using IAX2 and the other using SIP. Well, for the  
past 2 days, I've noticed that the number of call drops in the group  
using SIP to connect the two servers has SIGNIFICANTLY reduced. For  
example, the group using IAX2 suffers from about 80-100 call drops a  
day, whereas the group using SIP suffers from about 20-30 call drops.  
This is from a sample of about 2000 calls per day.

It may or may not seem much, but is is a significant number of call  
drops. Even 20-30 is not acceptable, but it's improving.

I have asked for support from the list in the past regarding this  
issue but have not received much help. I even purchased a TDM400P  
board to put in box2 as a timer source to replace ztdummy but that  
hasn't helped.

I've done network troubleshooting as much as I can to no avail. All  
network interfaces show 0 collisions, 0 error packets, 0 CRC errors,  
0 packet loss. I have done random ping floods when agents experience  
call drops but everything turns out just fine.

All agents are using Windows-based PCs with eyeBeam as softphone.

The only "odd" thing I have seen is that when I run:

ping -i 0.1 -c 10000 {ip_address}

in order to almost flood ping an agent's station from the box2  
asterisk server, the average latency is .2ms but every 1000 packets  
(approx), the latency jumps to 1ms. I would assume this would, at  
worst, affect call quality (jitter) but not a call drop.

 From a post I read this weekend, someone mentioned that maybe one  
(or more) of the PCs could be infected with a virus which could be  
causing this. I've been tempted to format a few machines with Linux  
and X-Lite to see if that improves.

BTW, I've tried using IAX-based softphones and the experience is  
still similar with regards to the call drops.

box2 is a Dual-Xeon 2.4GHz, 2GB RAM, 3ware SATA RAID 5, using the  
onboard Intel 10/100/1000 Etherexpress NIC, and handles merely 25  
agents. I manually configured the interrupts to make sure CPU1  
handles all TDM400P interrupts while CPU0 handles the rest (as some  
of the documentation suggested) but problem still persists.

I don't do anything fancy in Asterisk. Answer a call, put the call on  
queue, deliver to agents using rrmemory strategy. That's all. I was  
even thinking of getting an account with a VoIP provider in order to  
divert some traffic there and see if asterisk connecting directly  
with a VoIP provider will handle my setup better. From what I've read  
on the list, I have a very very very simple setup and it doesn't make  
any sense for me to be getting this many call drops. I'm simply  
running out of ideas and almost ready to throw in the towel.

I hope this has answered your curiosity and hopefully will allow you  
or others in helping me troubleshoot this issue further.

Thanks,
Waldo

On Dec 6, 2005, at 10:22 PM, Alvaro Parres wrote:

> Why using SIP instead of IAX2 ???
>
> Only a question becouse i always use IAX
>
>
>
>
> On 12/6/05, Waldo Rubinstein <waldo at trianet.net> wrote:
> Well... not so perfectly.
>
> What I'm experiencing is that during certain call volumes, many calls
> go thru from box1 to box2. However, there are some cases where I get
> this message:
>
> Dec  6 11:11:19 WARNING[203]: chan_sip.c:9525 handle_response_invite:
> Forbidden - wrong password on authentication for INVITE to
> '"5095551212" <sip:5095551212 at 10.0.0.1>;tag=as3e387d65'
>
> and the caller gets busy signal. However, other callers calling the
> same number get thru with no problems. Why is this?
>
> Thanks,
> Waldo
>
> On Dec 5, 2005, at 10:30 AM, Waldo Rubinstein wrote:
>
> > This worked perfectly.
> >
> > Thanks,
> > Waldo
> >
> > On Dec 5, 2005, at 4:32 AM, xcel wrote:
> >
> >>
> >> Try this
> >>
> >> ___________________________________
> >> 1st Machine sip.conf
> >>
> >> [box2]
> >> username=box1
> >> type=friend
> >> host= 10.0.0.2
> >> secret=*****
> >>
> >> in extensions.conf
> >>
> >> exten => _XXXXXX,1,Dial(SIP/box2/${EXTEN})
> >>
> >> __________________________________
> >> 2nd Machine sip.conf
> >>
> >> [box1]
> >> username=box2
> >> type=friend
> >> host=10.0.0.1
> >> secret=*****
> >>
> >> in extensions.conf
> >> exten => _XXXXX,1,Dial(SIP/box1/${EXTEN})
> >>
> >> --xce
> >>
> >>
> >> *********** REPLY SEPARATOR  ***********
> >>
> >> On 12/5/2005 at 12:11 AM Waldo Rubinstein wrote:
> >>
> >>> I have 2 Asterisk servers running 1.2.0. One of them is a PSTN
> >>> gateway. Currently they are connected using IAX2. I wanted to play
> >>> with SIP.
> >>>
> >>> I setup a sip entry (type=friend) in the PSTN gateway box and a  
> sip
> >>> entry (type=user) in the second box in order to send calls  
> using SIP
> >>> to the second box. This works fine. However, when I setup the  
> second
> >>> box as type=friend in order for it to be able to send calls  
> back to
> >>> the gateway box, then calls no longer work from gateway box to the
> >>> second box. The reported error is:
> >>>
> >>> Dec  5 00:07:14 NOTICE[203]: chan_sip.c:9514  
> handle_response_invite:
> >>> Failed to authenticate on INVITE to '"2125551212" <sip:
> >>> 2125551212 at 10.0.10.1>;tag=as0698b1b9'
> >>>
> >>> In the gateway box, my sip.conf looks like this:
> >>>
> >>> [general]
> >>> allowguest=yes
> >>> autocreatepeer=no
> >>>
> >>> [secondbox]
> >>> type=friend
> >>> host= 10.0.0.2
> >>> secret=mysecret
> >>>
> >>> In the second box, my sip.conf looks like this:
> >>>
> >>> [general]
> >>> allowguest=yes
> >>> autocreatepeer=no
> >>>
> >>> [secondbox]
> >>> type=user
> >>> host=10.0.0.1
> >>> secret=mysecret
> >>>
> >>> Any ideas on how to correctly set this up?
> >>>
> >>> Thanks,
> >>> Waldo
> >>> _______________________________________________
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> >>>
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> >>
> >>
> >>
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> >
>
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