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Dropping VoicePulse Registration

JoeB
Posts: 2
 Member Since:
2006-11-28
Submitted by JoeB on Mon, 04/28/2008 - 6:31am.

Hello,

I have been using VoicePulse for about a year now and until recently have not had any problems. Starting about a month ago the registration with VoicePulse drops and we lose the ability to make outbound calls and receive inbound calls.

Unfortunately since we receive mostly inbound calls it may be hours before we realize that our clients cannot call in. If I do a sip reload or stop and start amportal it registers and calls come in fine again.

1) What am I looking for in the logs to identify this problem. I've scanned through the full log but don't know what to look for. I also don't have an exact time that we are losing registration.

2) Is this a known issue. I tried searching in the forums but searching for sip registration and problem returns a lot of results.

Trixbox 2.4.0
VoicePulse Connect
Cisco 7960s

Thanks,
Joe

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rmcconky
Posts: 18
 Member Since:
2006-07-09
Same issue
Tue, 04/29/2008 - 10:51am

I am seeing the same issue on a customers system. They will drop a call, then nothing. The SIP registry shows Auth Sent and never changes. A sip reload fixes it until it happens again.



rmcconky
Posts: 18
 Member Since:
2006-07-09
Ticket
Tue, 04/29/2008 - 10:43pm

I have logged a ticket with VP and bumped up the logging on the server. I would try and get some packet captures, but it is so random it would take up a lot of space letting it run indefinitely.



dswartz
Posts: 142
 Member Since:
2006-09-26
here's a thought:
Wed, 04/30/2008 - 5:48am

use the asterisk API to have a cron job do a 'sip show registry' and grep for the string 'Auth Sent' and if you see that, wait a few seconds and retry and if still that state, restart?



dswartz
Posts: 142
 Member Since:
2006-09-26
here's a thought:
Wed, 04/30/2008 - 5:48am

use the asterisk API to have a cron job do a 'sip show registry' and grep for the string 'Auth Sent' and if you see that, wait a few seconds and retry and if still that state, restart?



dswartz
Posts: 142
 Member Since:
2006-09-26
here's a thought:
Wed, 04/30/2008 - 5:50am

use the asterisk API to have a cron job do a 'sip show registry' and grep for the string 'Auth Sent' and if you see that, wait a few seconds and retry and if still that state, restart?



JoeB
Posts: 2
 Member Since:
2006-11-28
dswartz - that's similar to
Wed, 04/30/2008 - 8:41am

dswartz - that's similar to what I've been thinking as a temporary fix.



techmon
Posts: 6
 Member Since:
2006-09-14
dswartz or JoeB, Would you
Wed, 04/30/2008 - 9:57am

dswartz or JoeB,

Would you be sol kind as to give me the exact shell command in order to perform that?



dswartz
Posts: 142
 Member Since:
2006-09-26
I can't say offhand
Wed, 04/30/2008 - 1:42pm

since I've never done this specific task, but the principle should be sound. i'd recommend googling for the asterisk API (perl and/or php are available). something here might be helpful:

http://www.voip-info.org/wiki/view/Asterisk+manager+API

Example (you would write a php/perl script to do this): I did telnet to localhost on port 5038 and:

[root@sphinx ~]# telnet localhost 5038
Trying 127.0.0.1...
Connected to localhost.
Escape character is '^]'.
Asterisk Call Manager/1.0
Action: login
Username: admin
Secret: XXXXXX

Response: Success
Message: Authentication accepted

Action: sippeers

(a ton of crap will print)

if you stay connected, you should see periodic messages like:

Event: Registry
Privilege: system,all
ChannelDriver: SIP
Domain: chiv1.voipstreet.com
Status: Registered

You want to look at the "Status:" field and based on actions off of that...



dswartz
Posts: 142
 Member Since:
2006-09-26
what might be easier
Wed, 04/30/2008 - 3:17pm

is to try floAPI.php. google for where to download it. it's a class library that builds in a lot of the nuts&bolts for using asterisk manager interface.



dswartz
Posts: 142
 Member Since:
2006-09-26
what might be easier
Wed, 04/30/2008 - 3:19pm

is to try floAPI.php. google for where to download it. it's a class library that builds in a lot of the nuts&bolts for using asterisk manager interface.



rmcconky
Posts: 18
 Member Since:
2006-07-09
Talked to VP today
Wed, 04/30/2008 - 10:44pm

One of their tech support people called me and said they found and resolved an issue with Asterisk on their end. They did reccomend the above work around if it did not resolve it.

My issue was I have had Viatalk trunks for 3 years now and have never had SIP issues with them.

Thus far in the past 24 hours no registration issues.



acetechgroup
Posts: 8
 Member Since:
2008-05-01
i've been having problems
Thu, 05/01/2008 - 6:55am

i've been having problems with Voicepulse for a month or so; they seem to drop us every friday (and sometimes more often than that).

i inquired and found out that their technicians are "doing upgrades" (i.e. breaking, fixing, then re-breaking) things often.

it's this kind of unreliable service that is making me switch to bandwidth.com as a provider.
i've been immensely more impressed by the quality of professionalism and customer care they provide.



dgoner
Posts: 111
 Member Since:
2006-05-31
I've been noticing
Fri, 05/02/2008 - 5:14pm

I've been noticing disconnects on our Voicepulse trunks for the past 2 weeks. The trunks had been reliably working for months. It's really gotten annoying lately as they are happening every couple of days. I don't think they have fixed the issue because this happened again today. I have yet to issue a ticket but have been thinking about giving bandwidth.com a whirl.



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