search for: defaultus

...ed my sip.conf so: register => 00493511111111:MYSECRET at pbxluca/00493511111111 register => 00493512222222:MYSECRET at pbxfax/00493512222222 register => 00493513333333:MYSECRET at pbxanika/00493513333333 register => 4444444444:MYVERYSECRET at messagenet/4444444444 [pbxluca] type=peer defaultuser=00493511111111 secret= MYSECRET dtmfmode=rfc2833 host=172.16.34.132 context=luca_incoming outboundproxy=172.16.34.132 port=5060 fromuser=00493511111111 fromdomain=172.16.34.132 usereqphone=yes canreinvite=no insecure=invite qualify=yes qualifyfreq=600 [pbxfax] type=peer defaultuser=004935122222...

Hi All, I am running Asterisk 1.6.1.2 with realtime for SIP and recently noticed the values in the defaultuser field have been "disappearing". I can't place my finger on what is happening but it appears that when the peer de-registers the defaultuser field is cleared. I will be having a more detailed look through the logs and possibly add more logging to the database but wondered if anyone...

1.6.2.13, sip.conf: [155] type=friend context=longdistance callerid="Admin" <155> secret=test host=dynamic dtmfmode=rfc2833 allow=all defaultuser=155-trust ............ On aastra: Basic SIP Authentication Settings Screen Name Phone Number 155 Caller ID 155 Authentication Name 155-trust Password test But: WARNING[1737]: chan_sip.c:12800 check_auth: username mismatch, have <155>, digest...

...ne I keep getting this error on my log files: [2014-02-17 19:55:18] WARNING[20569] res_config_sqlite3.c: Could not execute 'UPDATE "sip_buddies" SET "ipaddr" = '192.168.2.23', "port" = '5060', "regseconds" = '1392692118', "defaultuser" = '2012', "useragent" = 'Aastra 57i/3.2.2.2044', "lastms" = '-1', "fullcontact" WHERE "name" = '2012'': near "2012": syntax error [2014-02-17 19:55:30] WARNING[20569] res_config_sqlite3.c: Could not ex...

....co.uk:5060/sj0151 register => user7:password7 at callcentric.com:5060/sj1777 register => user8:password8 at sipgate.co.uk:5060/sj0203 I then have a selection of #included files. The first defines se2489: [se2489] type=friend insecure=port,invite secret=password1 defaultuser=user1 fromuser=user1 fromdomain=sipgate.co.uk host=sipgate.co.uk port=5060 outboundproxy=proxy.live.sipgate.co.uk disallow=all allow=ulaw context=external-se qualify=yes canreinvite=no dtmfmode=rfc2833 The second defines sj0151: [sj0151] type=...

...wrote in my sip.conf: register => 00493511111111:MYSECRET at pbxluca/00493511111111 register => 00493512222222:MYSECRET at pbxfax/00493512222222 register => 00493513333333:MYSECRET at pbxanika/00493513333333 register => 4444444444:MYSECRET at messagenet/4444444444 [pbxluca] type=peer defaultuser=00493511111111 secret= MYSECRET dtmfmode=rfc2833 host=172.16.34.132 context=luca_incoming outboundproxy=172.16.34.132 port=5060 fromuser=00493511111111 fromdomain=172.16.34.132 usereqphone=yes canreinvite=no insecure=invite [pbxfax] type=peer defaultuser=00493512222222 secret= MYSECRET dtmfmode=...

Hi list! Finally I got my Asterisk works with my two phones... It was a problem on my Firewall (for the phone of my wife) and on my Dialplan (for forwarding calls). Now all works as expected, at least in the simulation I did with AsteriskNOW. Hopefully it will work later, when Deutsche Telekom changes my ISDN to VoIP... Well, now I have some time to spend with "fooling"... My phone

...> <--- Reliably Transmitting (NAT) to 204.11.192.161:5060 ---> > SIP/2.0 401 Unauthorized asterisk is trying to find a peer based on the _calling number_! Here's the callcentric peer based on its support pages: [callcentric] type=peer context=from-callcentric host=callcentric.com defaultuser=1777<myccid> ;defaultuser=1914<mycctelnum> secret=<mysecret> fromuser=1777<myccid> ;fromuser=1914<mycctelnum> fromdomain=callcentric.com ;fromdomain=ss.callcentric.com insecure=port,invite disallowed_methods=UPDATE directmedia=no videosupport=no disallow=all allow=ul...

...> 00493511111111:MYSECRET at pbxluca/00493511111111 > register => 00493512222222:MYSECRET at pbxfax/00493512222222 > register => 00493513333333:MYSECRET at pbxanika/00493513333333 > register => 4444444444:MYSECRET at messagenet/4444444444 > > [pbxluca] > type=peer > defaultuser=00493511111111 > secret= MYSECRET > dtmfmode=rfc2833 > host=172.16.34.132 > context=luca_incoming > outboundproxy=172.16.34.132 > port=5060 > fromuser=00493511111111 > fromdomain=172.16.34.132 > usereqphone=yes > canreinvite=no > insecure=invite > > [pbxfax]...

..."0473990000" <sip:0473990000 at 91.208.12.134>;tag=as6c1c6d30 329909006666 = SIP username @ provider 3291119600 = number that I call 0473990000 = my number that I'm calling with sip.conf settings : [329909006666] type=friend host=ssw.brussels.provider username=329909006666 defaultuser=329909006666 fromuser=329909006666 secret=wknxxyav6e context=PROVIDERin disallow=all allow=alaw allow=gsm qualify=yes canreinvite=no dtmfmode=rfc2833 amaflags=billing What setting do I need to change in asterisk 1.6.2.11 ?? Kind regards, Jonas. -------------- next part -------------- An HTML...

Hi, We face a strange behavior with Asterisk 1.8.15 and SIP defaultuser definition. in sip.conf [blabla0](natted-phone,ulaw-phone,callgroup1,snom-320) defaultuser=tel-221 mailbox=221 callerid="My CID" dtmfmode=auto ;defaultip=10.0.12.21 CLI sip show peer blabla0 Addr->IP : 10.0.12.21:2067 Defaddr->IP : (null) Prim.Trans...

...on my logs: [Apr 29 13:09:32] WARNING[30494]: res_config_sqlite3.c:520 realtime_sqlite3_execute_handle: Could not execute 'UPDATE "sip_buddies" SET "ipaddr" = '192.168.0.52', "port" = '5060', "regseconds" = '1398795032', "defaultuser" = '112', "useragent" = 'Zoiper r21999', "lastms WHERE "name" = '112'': near "112": syntax error5060;rinst [Apr 29 13:10:26] WARNING[30494]: res_config_sqlite3.c:520 realtime_sqlite3_execute_handle: Could not execute 'UPDAT...

...gs include: [local-phone](!) context=mylocal type=friend nat=no canreinvite=no host=dynamic qualify=yes dtmf=info language=fr call-limit=5 subscribecontext=subs disallow=all allow=alaw t38pt_udptl=no setvar=accountcode=foo [168](local-phone) defaultuser=168 secret=pass168 callerid=John Doe<168> mailbox=168 setvar=longcid=015555555 setvar=accountcode=bar CLI> sip show peer 168 ... Variables : accountcode = bar longcid = 015555555 accountcode = foo When running, ${SIPPEER(168,cha...

...up=yes language=de register => 00493511111111:MYSECRET at pbxluca/00493511111111 register => 00493512222222:MYSECRET at pbxfax/00493512222222 register => 00493513333333:MYSECRET at pbxanika/00493513333333 register => 4444444444:MYVERYSECRET at messagenet/4444444444 [pbxluca] type=peer defaultuser=00493511111111 secret= MYSECRET dtmfmode=rfc2833 host=172.16.34.132 context=luca_incoming outboundproxy=172.16.34.132 port=5060 fromuser=00493511111111 fromdomain=172.16.34.132 usereqphone=yes canreinvite=no insecure=invite [pbxfax] type=peer defaultuser=00493512222222 secret= MYSECRET dtmfmode=...

...asterisk". Is this a bug? Or is there some other config I must make ? register = 2155551941:123456 at 10.0.138.226/2155551941~600 [peer](!) type=peer context=inbound qualify=yes qualifyfreq=300 insecure=port,invite nat=yes outgoinglimit=4 incominglimit=4 [mypeer](peer) host=10.0.138.226 defaultuser=2155551941 fromuser=2155551941 md5secret=023f30a320a5781e8ffd1af9888012af incominglimit=10 IP (tos 0x0, ttl 64, id 9242, offset 0, flags [none], proto UDP (17), length 555) 10.0.1.3.5060 > 10.0.138.226.5060: SIP, length: 527 OPTIONS sip:10.0.138.226 SIP/2.0 Via: SIP/2.0/UD...

Hi, I am using Asterisk currently as a SIP proxy to a SIP provider. Inside my real-time, I would set the SIP credential based on what the user has provided. For example [name] type=peer defaultuser=USER_PROVIDED secret=USER_PROVIDED host=USER_PROVIDED When I reset Asterisk, Asterisk will attempt to register with the sip provider. And if there are sufficiently amount of records with invalid credentials, I'll get blocked by the SIP provider as they might think that I'm brute forcing....

...indaddr = 0.0.0.0 context=default qualify=no disallow=all allow=alaw allow=ulaw allow=g729 allow=gsm allow=slinear srvlookup=yes videosupport=yes alwaysauthreject=yes register => SIP-ID:SIP-Password at sipgate.co.uk/SIP-ID [sipgate] type=peer secret=SIP_PASSWORD insecure=invite username=SIP-ID defaultuser=SIP-ID fromuser=SIP-ID context=sipgate_in fromdomain=sipgate.co.uk host=sipgate.co.uk outboundproxy=proxy.live.sipgate.co.uk qualify=yes disallow=all allow=alaw dtmfmode=rfc2833 SIP-ID:SIP-Password obviously, i replace these with my login details but, are these the same thing ? SIP-Password SI...

Hi I have some question about mysql realtime. 1) Anyone know exactly if there is a specific order to declare sip table column for realtime ? In which file can I find that order ? 2) In my extconfig.conf, [settings] are : sipusers => mysql,general,siptable sippeers => mysql,general,siptable so means that I use realtime dynamic exactly ? Is it normal if some parameters from sip.conf still

Kevin Larsen <kevin.larsen at pioneerballoon.com> schrieb: > The phone you gave your wife is really old. Are you sure it supports SIP > OPTIONS? Can you make a call in or out to it? If you can, it is more > likely that it just doesn't support that and you can't use a qualify > statement. No, I'm not sure. And no, I can't make any call, right now... At least,

...like the phone is not registered. The phone and the asterisk are in the same local network. On asterisk side: Cawdor*CLI> sip show peers ... 55/55 (Unspecified) D N 5060 UNKNOWN ... What am I missing? Best regards, Lukasz sip.conf: [55] type=friend defaultuser=55 secret=12345655 context=home_castle callerid="Lukasz Cisco 7960" <55> canreinvite=no host=dynamic dtmfmode=rfc2833 qualify=200 mailbox=55 SIPDefault.cnf: image_version: P0S3-8-12-00 proxy1_address: 192.168.1.109 ; Can be dotted IP or FQDN proxy_register: 1 messages_uri: 80...