Re: [asterisk-users] Asterisk removes a charachter from sip peer name
Perfect, that's it! Thank you Paddy for pointing that out to me, I had
totally missed it!
Thanks,
Olli
2015-01-05 15:15 GMT+02:00 Paddy Grice :
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Olli Heiskanen
> *Sent:* 03 January 2015 08:04
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* [asterisk-users] Asterisk removes a charachter from sip peer
> name
>
>
>
> Hello all,
>
> Just wondering on a behavior I noticed while testing with realtime sip
> peers with names like 111@mydomain.com. Using Kamailio as outbound
> proxy, it sends Asterisk a sip message where To header value is <
> sip:111@mydomain.com> and From header has value "username" <
> sip:111@mydomain.com;transport=UDP>;tag=fc609171. When Asterisk sends
> out the sip message, the To header is as it was but as for From header,
> Asterisk removes the "." charachter from the user part of the sip uri, thus
> resulting in 111333. Also the Contact header is affected the same way.
>
> I was wondering what might be causing this? Does Asterisk not allow dots
> in the peer names? The call itself connects so it's not much of an issue
> but it would be good to know about this, as of course there's a chance I've
> just missed something relevant.
>
> cheers,
> Olli
>
> Sounds a bit like
>
> From sip.conf
>
> ; The shrinkcallerid function removes '(', ' ', ')', non-trailing '.', and
> '-' not
> ; in square brackets. For example, the caller id value 555. becomes
> 555
> ; when this option is enabled. Disabling this option results in no
> modification
> ; of the caller id value, which is necessary when the caller id represents
> something
> ; that must be preserved. This option can only be used in the [general]
> section.
> ; By default this option is on.
> ;
> ;shrinkcallerid=yes ; on by default
> Paddy
>
>
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Re: [asterisk-users] Asterisk removes a charachter from sip peer name
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olli Heiskanen
Sent: 03 January 2015 08:04
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Asterisk removes a charachter from sip peer name
Hello all,
Just wondering on a behavior I noticed while testing with realtime sip peers
with names like 111@mydomain.com. Using Kamailio as outbound proxy, it
sends Asterisk a sip message where To header value is
mailto:sip%3a111@mydomain.com> > and From
header has value "username" mailto:sip%3a111@mydomain.com> ;transport=UDP>;tag=fc609171. When
Asterisk sends out the sip message, the To header is as it was but as for
>From header, Asterisk removes the "." charachter from the user part of the
sip uri, thus resulting in 111333. Also the Contact header is affected the
same way.
I was wondering what might be causing this? Does Asterisk not allow dots in
the peer names? The call itself connects so it's not much of an issue but it
would be good to know about this, as of course there's a chance I've just
missed something relevant.
cheers,
Olli
Sounds a bit like
>From sip.conf
; The shrinkcallerid function removes '(', ' ', ')', non-trailing '.', and
'-' not
; in square brackets. For example, the caller id value 555. becomes
555
; when this option is enabled. Disabling this option results in no
modification
; of the caller id value, which is necessary when the caller id represents
something
; that must be preserved. This option can only be used in the [general]
section.
; By default this option is on.
;
;shrinkcallerid=yes ; on by default
Paddy
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[asterisk-users] Asterisk removes a charachter from sip peer name
Hello all, Just wondering on a behavior I noticed while testing with realtime sip peers with names like 111@mydomain.com. Using Kamailio as outbound proxy, it sends Asterisk a sip message where To header value is < sip:111@mydomain.com> and From header has value "username" < sip:111@mydomain.com;transport=UDP>;tag=fc609171. When Asterisk sends out the sip message, the To header is as it was but as for From header, Asterisk removes the "." charachter from the user part of the sip uri, thus resulting in 111333. Also the Contact header is affected the same way. I was wondering what might be causing this? Does Asterisk not allow dots in the peer names? The call itself connects so it's not much of an issue but it would be good to know about this, as of course there's a chance I've just missed something relevant. cheers, Olli -- _ -- 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
