search for: spip_450

...p:0100@<CENSORED>;user=phone> CSeq: 2 INVITE Call-ID: d2c5011e-1d718f7-9203b936@<CENSORED POLYCOM IP> Contact: <sip:3109@<CENSORED POLYCOM IP>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_450-UA/4.0.7.2514 Accept-Language: en Supported: 100rel,replaces Allow-Events: conference,talk,hold Authorization: Digest username="3109", realm="asterisk", nonce="<CENSORED>", uri="sip:0100@<CENSORED>:5060;user=phone", response="<CENSORED>...

...user is in a call and presses the transfer soft key, they get dial tone, and start typing the extension, say 1008. But by the time they get 100 typed in, the phone tries to dial and the transfer fails. I feel sure that it's a dial plan issue on the phone. We are running: PolycomSoundPointIP-SPIP_450-UA/3.3.3.0094 The dialplan section of the sip.cfg provisioning file is: <dialplan dialplan.1.impossibleMatchHandling="0" dialplan.1.removeEndOfDial="1" dialplan.1.applyToUserSend="1" dialplan.1.applyToUserDial="1" dialplan.1.applyToCallListDial="...

...2.168.1.70 Port 5060 Defaddr->IP : 0.0.0.0 Port 5060 Transport : UDP Def. Username: Barry's IP450 SIP Options : (none) Codecs : 0x4 (ulaw) Codec Order : (ulaw:20) Auto-Framing : No 100 on REG : No Status : OK (14 ms) Useragent : PolycomSoundPointIP-SPIP_450-UA/3.1.3.0439 Reg. Contact : sip:6668 at 192.168.1.70 Qualify Freq : 60000 ms Sess-Timers : Accept Sess-Refresh : uas Sess-Expires : 1800 secs Min-Sess : 90 secs - From what I have read, with 'Busy Level = 1' I should be seeing BUSY returned from the DEVICE_STATE() call,...

...p:0100@<CENSORED>;user=phone> CSeq: 2 INVITE Call-ID: d2c5011e-1d718f7-9203b936@<CENSORED POLYCOM IP> Contact: <sip:3109@<CENSORED POLYCOM IP>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_450-UA/4.0.7.2514 Accept-Language: en Supported: 100rel,replaces Allow-Events: conference,talk,hold Authorization: Digest username="3109", realm="asterisk", nonce="<CENSORED>", uri="sip:0100@<CENSORED>:5060;user=phone", response="<CENSORED>...

Hello, fresh install of Asterisk 13.11.2, client unable to register. For now I have IPtables disabled, also selinux is disabled [1006] type=friend username=1006 secret=mysecret context=sip-phone call-limit=1 callerid="iuser" <1006> disallow=all host=dynamic allow=all any ideas? Thanks, Motty -------------- next part -------------- An HTML attachment was

...n.com CSeq: 1 REGISTER Call-ID: 6cbe37bb-cca69d70-85d0431d at 192.168.1.37 Contact: <sip:1006 at 192.168.1.37:5060>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER" User-Agent: PolycomSoundPointIP-SPIP_450-UA/4.0.10.0689 Accept-Language: en Max-Forwards: 70 Expires: 90 Content-Length: 0 Sip.conf [1006] type=friend username=1006 secret=mysecret context=sip-phone call-limit=5 callerid="iuser" <1006> disallow=all host=dynamic allow=all...