Hi Guys,
I have observed that 46XX cvonverted to SIP can't dial numbers starting with * characters, for example number *077777.
I have tried with others hardphone or softphone and it works correctly
Phone is registered with Asterisk server after looking into traces when dialling a number starting with * the domain is not added after the number into the SIP INVITE, see below
See below an example of a call when I dial from Avaya phone the number *077777:
See below an example when I dial number 40075 domain name is correctly indicated into the INVITE:
so it really looks to be related to the way how Avaya sip firmware handle number starting with *.
Let me know if you see a way to resolve it or if it is definitively a bug from Avaya phone SIP firmware.
Best Regards.
I have observed that 46XX cvonverted to SIP can't dial numbers starting with * characters, for example number *077777.
I have tried with others hardphone or softphone and it works correctly
Phone is registered with Asterisk server after looking into traces when dialling a number starting with * the domain is not added after the number into the SIP INVITE, see below
See below an example of a call when I dial from Avaya phone the number *077777:
Code:
<--- SIP read from UDP:10.147.116.240:5060 ---> INVITE sip:*077777 SIP/2.0 Via: SIP/2.0/UDP 10.147.116.240:5060;branch=z9hG4bKac0f54add Max-Forwards: 70 Content-Length: 268 [COLOR="Red"]To: *077777 <sip:*077777>[/COLOR] From: 46044 <sip:[email protected]>;tag=e720c41b0ffa03b Call-ID: [email protected] CSeq: 1821799987 INVITE Route: <sip:10.147.113.73;lr> Supported: timer Allow: NOTIFY Allow: REFER Allow: OPTIONS Allow: INVITE Allow: ACK Allow: CANCEL Allow: BYE Content-Type: application/sdp Contact: 46044 <sip:[email protected]:5060> Supported: replaces User-Agent: Avaya SIP R2.2 Endpoint Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 v=0 o=MxSIP 0 2143443207 IN IP4 10.147.116.240 s=SIP Call c=IN IP4 10.147.116.240 t=0 0 m=audio 34008 RTP/AVP 0 8 18 2 127 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:127 telephone-event/8000 a=ptime:20 <-------------> --- (21 headers 12 lines) --- Sending to 10.147.116.240:5060 (NAT) Using INVITE request as basis request - [email protected] Found peer '46044' for '46044' from 10.147.116.240:5060 <--- Reliably Transmitting (no NAT) to 10.147.116.240:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.147.116.240:5060;branch=z9hG4bKac0f54add;received=10.147.116.240 From: 46044 <sip:[email protected]>;tag=e720c41b0ffa03b To: *077777 <sip:*077777>;tag=as4a3c298e Call-ID: [email protected] CSeq: 1821799987 INVITE Server: DTC Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="69023bd1" Content-Length: 0
See below an example when I dial number 40075 domain name is correctly indicated into the INVITE:
Code:
<--- SIP read from UDP:10.147.116.240:5060 ---> INVITE sip:[email protected] SIP/2.0 Via: SIP/2.0/UDP 10.147.116.240:5060;branch=z9hG4bK177745ee3 Max-Forwards: 70 Content-Length: 268 [COLOR="red"]To: 40075 <sip:[email protected]>[/COLOR] From: 46044 <sip:[email protected]>;tag=525fc3f33445702 Call-ID: [email protected] CSeq: 328660852 INVITE Route: <sip:10.147.113.73;lr> Supported: timer Allow: NOTIFY Allow: REFER Allow: OPTIONS Allow: INVITE Allow: ACK Allow: CANCEL Allow: BYE Contact: 46044 <sip:[email protected]:5060> Content-Type: application/sdp Supported: replaces Authorization:Digest response="22e530e0f093ae7d6804c9ceb91410dd",username="46044",realm="asterisk",nonce="20d0f315", algorithm=MD5,uri="sip:[email protected]" User-Agent: Avaya SIP R2.2 Endpoint Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26
Let me know if you see a way to resolve it or if it is definitively a bug from Avaya phone SIP firmware.
Best Regards.
