similar to: Odd SIP error message

On about 25% of inbound calls to a ring group, picking up any one extension as it rings results in dead air. Some details regarding my VoIP network to make the following logs more readable: 192.168.7.130 resolves to the trixbox host. 192.168.7.135 resolves to endpoint 812. 192.168.7.137 resolves to endpoint 811. 192.168.7.138 resolves to endpoint 810. 192.168.7.139 resolves to endpoint 813.

How do I clear a global variable for good? I have a situation of needing to use global variables to aide in channel communication, but will be changing the name within a defined scope. Additional Background... I want to get a variable from a channel (child) that is created by another channel (parent), however the execution of the parent channel does not continue until the child channel is gone.

Hi, the following logs are being generated while i test sip-to-sip windows software phones. Dec 7 17:05:16 WARNING[-159503440]: chan_sip.c:683 retrans_pkt: Maximum retries exceeded on call 40dedd1535853f17250b4d0854e35c17@200.75.243.237 for seqno 102 (Critical Request) == No one is available to answer at this time Dec 7 17:05:22 WARNING[-159503440]: chan_sip.c:683 retrans_pkt: Maximum

Did you try to boot without lan just the power ... I've had this same problem to and rebooted the device without lan connection -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Louis-David Mitterrand Sent: woensdag 26 januari 2005 11:36 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'

I'm having a problem with the callerID that the polycom IP600 phones are displaying. I would like to modify the CIDName and leave CIDNumber as exactly what the phone call came in as(provided they aren't hiding callerID). Most of the calls will be going to the queue, but a few will go directly to the SIP phones. I've done a various combinations of using SetCallerID(),

Hello, I have a polycom ip600 and eyebeam. When I call from polycom to eyeBeam, everything, including audio works. When I call the other side (from eyeBeam to polycom), I get no audio. In both cases, eyeBeam shows the same codec: g711u. Also sip show channels shows ulaw codec for both sides and correct addresses. I have canreinvite=no. I don't know if it's important, but asterisk

Hi list! Is anyone using the Kirk IP600/3 with SIP firmware connected to Asterisk? Any experiences / caveats? If anyone would be willing to share the dump of their IP600 config file, i would really appreciate it. Is there anything special i should put in my asterisk config? Thanks !!! Remco

Hello, After checking out CVS HEAD from yesterday (for those new PLC/Jitterbuffer patches), I was affected by bug 3795 with my Polycom IP600's. After seing it resolved as of this morning (thanks Mark), I decided to try again... I can answer incoming calls. No problem there. Putting calls on hold, however, results in my Polycom IP600 indicating the call on hold, but the caller does

I'm getting dozens of these at a very high rate: [Jan 20 09:15:27] NOTICE[16958]: chan_sip.c:12687 check_auth: Correct auth, but based on stale nonce received from '"" <sip:121 at gnat.com>;tag=as5f1a9480' [Jan 20 09:15:28] NOTICE[16958]: chan_sip.c:12687 check_auth: Correct auth, but based on stale nonce received from '"" <sip:130 at

I am using CVS-v1-0-04/20/05-12:31:26. Windows Messenger5.1 works MOH fine. After I stop MOH on Windows Messenger, if the hungup signal could not send to *, the sip channel(e.g.,SIP/52001-08ca) for this MOH remains. Then the user trys again MOH, a new sip channel starts. And again the hugup signal can not send to *,......... When I 'stop now' from CLI> , * cleanups the remaining sip

We have recently install an asterisk solution with about 60 physical extensions. While the system is running it runs reasonably well (Still have a few teething problems) but twice now they have experienced a degradation in voice quality and dropped calls and then finally asterisk completely crashes out. Restarting asterisk will work for a little while and it will crash again, each time less time

hi all i am using CVS with Realtime mysql on backend. Dial application is invoked again and again what is the reason. I have tested it by printing some message to debug. this application is invoked again and again here is debug you can see lot of messages from app_dial.c at the end. Any one tell me what is the reason. Is this a bug or what Kamran Ahmad

Hi, I have declared my sip users call-limit=2 and type=friend. When any user recieves a waiting call while already in a conversation, the peer call counter is set to 2.The problem is that, the counter is not reset to zero after hangup and becoz of this the user is not able to recieve any call anymore even if s/he has hungup. the asterisk cli displays the following error. [Mar 17 16:15:10]

Hi, I'm using SIP MESSAGE to asterisk V10 and it fails to be received. I'm not super sure of the reason but I'm making this guess: Due to I'm using non ipaddress in the to field, which contains sip:mobil1.xyz.com, Asterisk makes the mistake to try matching this name "mobil1.testserver.com" in extensions.conf and no extension/peer is found in the sip-message context

Hello everyone, This is is my first email to this group. I'm am writing a small php program to pull some info out of our Asterisk's queue_log. I'm having trouble figuring out what some of the parameters mean. Here's an example: 1118098465|1118098465.47|salesq|NONE|ENTERQUEUE||"Ray Balbin 25" <(716)250-3405> I found a doc that tells me about everything from

I am currently utilizing Polycom IP600 phones and presense. I have placed the hint directives in and everything seems to work fine. But this only works for a short period of time. After about 30 minutes, the extensions do not see when others are on the phone. Has anyone seen this type of behavior before? I am currently utilizing v1.5.2 of the polycom software and the cvs-head release from

Has anyone seen this one before? full:[Sep 14 11:49:01] DEBUG[15771] chan_sip.c: SIP attended transfer: Error: No target channel It coincided with a failed attended transfer... Ideas? PaulH

Have a strange problem... When a C7960 calls the Polycom ip600, the ip600's first line button blinks, the ip600 display shows the proper callerid, but the phone does not ring at all. If I call the same ip600 from a bt102, the ip600 rings properly. If I call the same ip600 from another C7960, the ip600 rings properly. All phones and asterisk are on the same lan within a few feet. The

I've noticed that when app_queue.so is reloaded(or just a reload command is used) that all queue members that were paused are automatically unpaused. Is there a workaround for this? (Note, I use statically defined callback agents). --johann

Hello all, I'm having a nightmare of a time trying to get stable results with SIP clients on Asterisk. I can't seem to find a configuration that works! In our office, we run a Sonicwall Pro 200, which is a sip aware, stateful firewall. Originally, I had configured Asterisk to run on the NAT side so that those within the office could connect easily, and those outside the office