Hello dear list. We run a Asterisk 1.6.2.6 on testbasis (SIP), and experience every day, when making calls, that the calls become silent. Not every calls, but 1 out of 3-4 calls, becomes silent suddenly during the conversation. When we then hangup, and redial immediately, the calls do not go through, we then have to try redial a couple of times, and then It suddenly gets through. There is nothing in the verbose log in Asterisk -r. SIP HW is Snom and Different types of Cisco. Anyone got an idea? Or at lest know how to dig deeper in logs? Med vennlig hilsen / Best regards Abacus IT AS - din Visma Software Partner - your Visma Software Partner L.Aksel Celasun Mobilnummer/cell phone: (+47) 900 15 103 Sentralbord/Support 4000 1850 aksel at abacus-it.no<mailto:aksel at abacus-it.no> Se denne m?nedens gode tilbud fra Abacus IT AS<http://www.abacus-it.no/systeml?sninger/kampanjer> -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20111018/b77d1088/attachment.htm>
I have similar problem at my home extension, but for that I know my phone's speaker is defective, and tapping it against the desk or wall fixes the problem. However in your case probably it is sip configuration (sip.conf or an included file), where canreinvite=yes where it should be canreinvite=no, either in general section, or in the extension settings. -- Zeeshan A Zakaria PBX - visionvoip.com Blog - ilovetovoip.com On 18/10/2011 09:35, Aksel Celasun wrote:> > Hello dear list. > > We run a Asterisk 1.6.2.6 on testbasis (SIP), and experience every > day, when making calls, that the calls become silent. > > Not every calls, but 1 out of 3-4 calls, becomes silent suddenly > during the conversation. > > When we then hangup, and redial immediately, the calls do not go > through, we then have to try redial a couple of times, and then It > suddenly gets through. > > There is nothing in the verbose log in Asterisk --r. > > SIP HW is Snom and Different types of Cisco. > > Anyone got an idea? Or at lest know how to dig deeper in logs? > > Med vennlig hilsen / Best regards > > Abacus IT AS > > - din Visma Software Partner > > - your Visma Software Partner > > *L.Aksel Celasun* > > Mobilnummer/cell phone: (+47) 900 15 103 > > Sentralbord/Support 4000 1850 > > aksel at abacus-it.no <mailto:aksel at abacus-it.no> > > Se denne m?nedens gode tilbud fra Abacus IT AS > <http://www.abacus-it.no/systeml%F8sninger/kampanjer> > > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20111018/f9321146/attachment.htm>
Aksel,
i faced a similar issue with remote sip extensions. and seems to be happening
due to internet problems. one way audio that is .. one of the parties (on site)
stops hearing the other party.
and it happens with one extension at a random timing and random extension.. and
if all extensions are on the same internet link it doesnt' happen to all of
them at once.. only one of them.
i suggest trying to change ISP for testing.
Tarek Sawah
Information Technology Adviser
Integrated Digital Systems
CCNP, MCSE, RHCE, TELECOM
USA: +1 386 492 9993
From: aksel at abacus-it.no
To: asterisk-users at lists.digium.com
Date: Tue, 18 Oct 2011 15:35:41 +0200
Subject: [asterisk-users] Problems during calls
Hello dear list. We run a Asterisk 1.6.2.6 on testbasis (SIP), and experience
every day, when making calls, that the calls become silent.Not every calls, but
1 out of 3-4 calls, becomes silent suddenly during the conversation.When we then
hangup, and redial immediately, the calls do not go through, we then have to try
redial a couple of times, and then It suddenly gets through.There is nothing in
the verbose log in Asterisk ?r. SIP HW is Snom and Different types of Cisco.
Anyone got an idea? Or at lest know how to dig deeper in logs? Med vennlig
hilsen / Best regardsAbacus IT AS- din Visma Software Partner- your Visma
Software Partner L.Aksel CelasunMobilnummer/cell phone: (+47) 900 15
103Sentralbord/Support 4000 1850aksel at abacus-it.no Se denne m?nedens gode
tilbud fra Abacus IT AS
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
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