[asterisk-users] Problems during calls

[Aksel Celasun]

Thank you for replying


My sip.conf is set to no on canreinvite



[general]
context=default
allowguest=yes
allowoverlap=no
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
pedantic=yes
tos_sip=cs3
tos_audio=ef
tos_video=af41
disallow=all
allow=alaw
;allow=ulaw
;allow=gsm
language=en
trustrpid = yes
sendrpid = yes
progressinband=never
useragent=TS200 PBX
promiscredir = no
usereqphone = no
dtmfmode = rfc2833
compactheaders = no
videosupport=no
maxcallbitrate=96
shrinkcallerid=yes
rtptimeout=60
rtpholdtimeout=300
rtpkeepalive=29
rtcachefriends=yes
recordhistory=yes
nat=yes
canreinvite=no
limitonpeers=yes
limitonpeer=yes
allowsubscribe=yes


Maybe there is something with the sip client, qualify=yes?

;Sentralbord
[501]
type=friend
secret=501
host=dynamic
context=phones
mailbox=501 at defualt
callerid=Sentralbord Abacus-IT
qualify=yes

Thank you in advance.

Regards

Aksel Celasun


Fra: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] På vegne av VisionVoIP
Sendt: 18. oktober 2011 15:49
Til: Asterisk Users Mailing List - Non-Commercial Discussion
Emne: Re: [asterisk-users] Problems during calls

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 månedens gode tilbud fra Abacus IT AS<http://www.abacus-it.no/systeml%F8sninger/kampanjer>






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