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Currently I am having 2 issues with my Avaya 4602 phone: First, the phone registers with my Asterisk server, but when I start dialing I get a busy signal after 4 digits. I specified in the dialplan on the phone to expect 10 digits and that solved that problem, but I still immediately get a busy after the 10th digit. The phone never sends a dial command to asterisk. Second, asterisk is

...e428be@192.168.0.254 CSeq: 102 INVITE From: "Bgate: Treatment (Large)" <sip:4005@192.168.0.254>;tag=as5b17ec6a To: <sip:mymobilenumber@192.168.0.253>;tag=2120bdca0a07567 Via: SIP/2.0/UDP 192.168.0.254:5060;branch=z9hG4bK6728b435 Content-Length: 0 User-Agent: MxSipApp/5.0.15.88 MxSF/v3.2.1.1 8 headers, 0 lines asterisk1*CLI> Sip read: INVITE sip:4000@192.168.0.254 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.253;branch=z9hG4bK2a550f4da Content-Length: 296 To: sip:4000@192.168.0.254 From: Incoming <sip:3330001@192.168.0.254>;tag=68b2ce27259fa46 Call-ID: 9dd4c369dbcf118b22b9...

Hello All, I am having an odd problem with Armin's chan-capi_cm on builds higher than 7955. It would seem that this happens on anything higher than 7955. What is happening is the isdn is ringing, then asterisk does a goto-if and just hangs. Asterisk itself is ok, but the isdn then rings out or busys out on the other side. Outgoing works fine, this only seems to effect incoming. I

...Call-ID: a85fb0e71a3e7def4bd7d7f158933c24 at 192.168.0.11 CSeq: 1006065354 REGISTER Contact: myname <sip:phone1 at 192.168.0.11:5060;transport=udp> Allow-Events: talk,hold,conference Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO User-Agent: Aastra 9133i/1.4.3.23 Brcm Callctrl/1.5.1.0 MxSF/v3.2.8.45 <-------------> --- (12 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 192.168.0.11 : 5060 (no NAT) <--- Transmitting (no NAT) to 192.168.0.11:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.11:5060;branch=z9hG4bKcd3373869;received=192...

...;branch=z9hG4bK667022457 Content-Length: 0 To: Port 3 <sip:3102@123.45.67.89> From: Port 3 <sip:3102@123.45.67.89>;tag=f8e5152d35870bf Call-ID: 9a610ba9dca9c942d8e2b12e89939fd3@123.45.67.89 CSeq: 1913617706 REGISTER Contact: Port 3 <sip:3102@0.0.0.0> User-Agent: MxSipApp/4.4.10.60 MxSF/v3.2.6.24 9 headers, 0 lines Using latest request as basis request Sending to 0.0.0.0 : 5060 (non-NAT) Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 0.0.0.0;branch=z9hG4bK667022457;received=98.76.54.32 From: Port 3 <sip:3102@123.45.67.89>;tag=f8e5152d35870bf To: Port 3 <sip:310...

...abac26fdac761f Call-ID: 3b4f38d8453c881116fdb6732d385ee4@10.144.169.136 CSeq: 1198861724 REGISTER Contact: 3000006666 <sip:3000006666@10.144.169.136> Allow: NOTIFY Allow: REFER Allow: OPTIONS Allow: INVITE Allow: ACK Allow: CANCEL Allow: BYE Content-Length: 0 User-Agent: Brcm Callctrl/1.5.0.0 MxSF/v3.2.5.20 16 headers, 0 lines Using latest request as basis request Sending to 10.144.169.136 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.144.169.136;branch=z9hG4bKc0bd6005d From: 3000006666 <sip:3000006666@10.144.166.220:5060>;tag=7abac26fdac761f To: 3000...

...NVITE Supported: timer Min-SE: 1800 Session-Expires: 3600 Allow: INVITE, ACK, BYE, CANCEL, REFER, NOTIFY Content-Type: application/sdp Contact: Port 1 <sip:21383396@192.168.0.27> Supported: replaces User-Agent: MxSipApp/4.4.13.88 MxSF/v3.2.7.38 Message body Frame 47 (537 bytes on wire, 537 bytes captured) Ethernet II, Src: 192.168.0.6 (00:0c:29:4e:99:37), Dst: 192.168.0.27 (00:90:f8:00:ef:d1) Internet Protocol, Src: 192.168.0.6 (192.168.0.6), Dst: 192.168.0.27 (192.168.0.27) User Datagram Protocol, Src Port: 5060 (5060), D...

....1.50 CSeq: 102 NOTIFY From: "Unknown" <sip:Unknown@192.168.1.50>;tag=as12a1c927 To: <sip:207@192.168.1.100>;tag=a7b3737f5e691cf Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK7032188a Content-Length: 0 Contact: <sip:207@192.168.1.100> User-Agent: Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 Sep ?8 18:44:31 VERBOSE[18779]: 9 headers, 0 lines Sep ?8 18:44:31 DEBUG[18779]: (Provisional) Stopping retransmission (but retaining packet) on '2680096f545a6d3701f95d6861b99f3e@192.168.1.50' Request 102: Found Sep ?8 18:44:31 VERBOSE[18779]: Sip read: SIP/2.0 200 OK Call-...

...GE Allow: INVITE Allow: ACK Allow: CANCEL Allow: BYE Allow: INFO Authorization:Digest response="2d5dd24c01e8db3a1ac1b918d471b1a0",username="cdot-109",realm="asterisk",nonce="7ec20f6d",uri="sip:209.139.212.169:5060" User-Agent: Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 =====OUT=====>>>>>>>>>>END SIP packet <<<<<<<<<<=====IN=====192.168.0.52: Received SIP packet from: 209.139.212.169:5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.52;branch=z9hG4bKe9dfdb692;received=69.90.106.130;rport=59...

...0379054bc60bad9c9e51b1579d46@192.168.153.100 CSeq: 1084359157 INVITE Supported: timer Allow: NOTIFY Allow: REFER Allow: OPTIONS Allow: INVITE Allow: ACK Allow: CANCEL Allow: BYE Content-Type: application/sdp Contact: sip:15800115@192.168.153.100 Supported: replaces User-Agent: Brcm Callctrl/1.5.1.2 MxSF/v3.2.6.26 and asterisk doesn't translate %23 to #. the grandstrem phones send it in this case: <-- SIP read from 192.168.50.224:5060: INVITE sip:#21@sip.tvnet.hu SIP/2.0 Via: SIP/2.0/UDP 192.168.50.224;branch=z9hG4bKba695b72a347ad40 From: <sip:15800101@sip.tvnet.hu>;tag=33c0bbf7cfe3...

...: NOTIFY, REFER, OPTIONS, INVITE, ACK, CANCEL, BYE Authorization:Digest response="48c35d36d1d994110615e715ef5ea23d",username="username",realm="asterisk",nonce="26a5abaa",algorithm=MD5,uri="sip:sip.provider.be:5060" User-Agent: Brcm-Callctrl/v1.7.1.1 MxSF/v3.6.2.5 And asterisk reply's with: SIP/2.0 403 Forbidden (Bad auth) Kenny

...From: "asterisk" <sip:asterisk@192.168.1.10>;tag=as5860bf17 To: <sip:avaya4602@192.168.1.98>;tag=cad443b1cd74b1e Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK69c2e97b Content-Length: 0 Contact: <sip:avaya4602@192.168.1.98> User-Agent: Elite 1.0 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 9 headers, 0 lines -- Got SIP response 481 "Call Does Not Exist" back from 192.168.1.98 Destroying call '7fc1849a4132a3a74565eeed3c26d507@192.168.1.10' Destroying call '3844dc0eeba9eb04adf132f539666c21@192.168.1.98'

Hi all, Im trying dial a user according to the device s/he uses. i mean if the user is using asterisk as a peer, then i have to pass the extension in the dial application like this: Dial(SIP/${EXTEN}@user) ;so that s/he can perform routing according to the DNID. and if the user is using sipura, linksys or grandstream i dial the user like this, Dial(SIP/user) so is there a way to know what kind

Hi, I've just (earlier today) updated from CVS so that I can apply the dtmf caller id patches. Unfortunately this has had an undesired effect. I have an intertex ix66 which up until the CVS update allowed me to register my * server with the ix66 for my local domain (eg sip.mydomain.com). Now it appears that asterisk gets totally confused and tries to register with itself! Anyone got any

Has anyone had any luck with PostgreSQL and Realtime? The realtime instructions on voip-info seem pretty straight forward... just not woking for me. I've included all of my config files below, and my console output. Entire console bootup output: [root@abox asterisk]# /usr/sbin/asterisk -vvvvvvc == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing