search for: shrinkcallerid

Am I really the only one having problems with this new "shrinkcallerid"? I can't find anything on Google about it. Was happening on 1.6.2.10 and now on 1.8.0-beta2 In sip.conf shrinkcallerid=no, yet a name like "Joe Smith" ends up being "JoeSmith" Whoever though this up anyway is stupid. Why would you want to strip spaces out of a call...

Hello all, Just wondering on a behavior I noticed while testing with realtime sip peers with names like 111.222 at mydomain.com. Using Kamailio as outbound proxy, it sends Asterisk a sip message where To header value is < sip:111.222 at mydomain.com> and From header has value "username" < sip:111.333 at mydomain.com;transport=UDP>;tag=fc609171. When Asterisk sends out the

...using 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 555.5555 becomes 5555555 ; when this option is enabled. Disabling this option results in no modification ; of the caller id value, which is...

...calleeid on calls but if the caller is between servers calleeid doesn't work. Callerid is working fine though. A call from a SIP client on asterisk2 to asterisk3. All phones are snom 760s. Any ideas or suggestions appreciated. iax.conf (asterisk2 10.6.1) [general] bandwidth=high allow=all shrinkcallerid=no [asterisk3] type=friend username=asterisk2 secret=secret host=10.101.0.3 context=incoming sendani=yes trunk=yes iax.conf (asterisk3 11.0.0-beta1) [general] bandwidth=high allow=all shrinkcallerid=no [asterisk2] type=friend username=asterisk3 secret=secret host=10.101.0.2 context=incoming s...

...uot; <2565551212>" So - I know the NAME field is getting into the system, but it's not showing up on the phones (and with telemarketers, that annoys my users). I'm using Asterisk 1.6.2.9, DAHDI 2.3.0 I have added callerid=asreceived to chan_dahdi.conf for my inbound trunks, and shrinkcallerid=no to my sip.conf. (without effect) Any ideas? THANKS Cassius

Hello again! I just got a SIP account and it seems - from a config on the net -, that I've configured it correctly. But I get no call to the outside. Registration was OK. I tried: channel originate sip/1/echo at iptel.org Application ... I see the channel active for a while, but no call gets established. In my config I have defined the section [iptel] for the outgoing call and I

...eep repeating and repeating but the call never actually takes place. The contents of my sip.conf file: ---------------------------------------------------------------------------------------------------- [general] context=unauthenticated allowguest=no srvlookup=no udpbindaddr=0.0.0.0 tcpenable=no shrinkcallerid=no [office-phone](!) type=peer context=LocalSets host=dynamic nat=force_rport,comedia dtmfmode=auto disallow=all allow=g729 [85004](office-phone) defaultuser=85004 secret=securepass callerid="Phone 4" <85004> [85014](office-phone) defaultuser=85014 secret=securepass callerid=&quo...