search for: callctrl

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

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

...g=24b6354e352ab62 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...

...20:5060>;tag=7abac26fdac761f 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=7abac26fdac...

...861b99f3e@192.168.1.50 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...

...IONS Allow: MESSAGE 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...

...06c Call-ID: 553a0379054bc60bad9c9e51b1579d46@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...

...;expires=300 Allow: 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

...CSeq: 102 NOTIFY 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

Hi all, We have a client that needs to setup about 80 desk phones (about 50 in one location and about another 30 in 5 different locations). Which brand/model would you recommend. We were personally thinking in recommending either Cisco, Aastra, Polycom, or Snom, for we've heard great things about them. However, having no real experience with them makes it hard in recommending one to

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