similar to: rtpmap encoding parameters & the 'unknown codec' problem?

Hello All, I have a media problem while using sip communicator user agent with asterisk behind NAT.I had enabled the debug mode in asterisk and capture the results.I have attached the results with this mail.Can any one help me to fix the problem? Thanks in advance, Partha __________________________________ Do you Yahoo!? Yahoo! Mail is new and improved - Check it out!

I've attached two SIP debugs in reference to bug #116. They are from today's CVS build. 1. pickup.txt is a call from SIP(1) to SIP(2) with SIP(3) picking up the call. After which, SIP(2) rings for about 30 seconds then stops. 2. hangup.txt is a call from SIP(1) to SIP(2) with SIP(1) hanging up before the call is answered. SIP(1&3) are Cisco 7960's and SIP(2) is a Polycom

I just downloaded the newest version from CVS(Tuesday@~7pm) and I am getting an error whenever I call the asterisk box. I cannot here any audio on the budgtone. This works fine with my pingtel phone and my sip 7960. Also if I call my Skinny 7960 it rings but I get that same error when I pick up. When the skinny phone calls the Budgtone it works fine. I have 2 budgtone phones and it does this on

Still no luck using asterisk as a proxy. 48 hours solid working on this. I'm beginning to think asterisk isn't going to be compatible with the provider I'm using :( Has anyone got *any* clues as to what can cause this message? It's definately provider specific (voiptalk works, pipecall doesn't) but confusingly seems to be caused by something in the client phone app. I

Hey all! I'm having problems trying to set up an ATA 186 with my Asterisk box. When I get the phone to place the call, I type the extension and I only get busy signal after 5 seconds. So I can't call my Asterisk box from my ATA and either call from my Asterisk to my ATA. Does anybody know what can be happing? Log is attached.. tks regards Oz -------------- next part -------------- >

I have spent the morning on this project, still without success. Summary: Yesterday I inadvertently unplugged my Grandstream phone. I might add I did a rebuild of my s/w from CVS at the same time. Since then, the Budgetone seems to talk SIP just fine, but the RTP being sent to it by asterisk "doesn't make any sound." It was suggested I do a factory reset of the phone, which I

The problem with the extra ptime descriptions in the SDP has been fixed in Asterisk (see http://lists.digium.com/pipermail/svn-commits/2006-October/017694.html). I've got the latest version of the 1.4 branch from SVN and have verified that the codec negotiation is working again. If you don't want to try the latest SVN version, then you'll have to restrict the phones to a single codec

I'm having a tough time getting call pickup to work on *. Here's my configuration: X100P with T-1, channels 1-4 voice <---> * <---CISCO 7960 with SIP 6.0 Image A call comes in, and * picks up and presents a menu. Caller chooses extension, (in this case ext 103, SIP/wsmith) Wsmith is sitting in my office, hears his phone ringing, picks up my phone, gets dial tone, and presses

Hello I am trying to connect linphone 0.12.2 to an * 0.9.1 box over a LAN using the console version of linphone. both boxs are using the latest alsa drivers on a LFS kernal 2.4. I am running into errors with codec compatability between linphone and *. A point to note is that I am able to connect to asterisk using other sip phones noteably sjphone however linephone is giving me

Hi, Could somebody help me this SIP trasport? I'm receiving SIP "invite" with CLI of calling party from the SIP gateway, aster that my IVR has to answer the call. sip.conf: ========= [general] port = 5060 bindaddr = 0.0.0.0 context = incomingsip videosupport=yes ; Turn on support for SIP video disallow=all ; Disallow all codecs allow=g729

Hi all, I am struggling to make my asterisk server work. The problem is I can not place a call from a phone outside, but I can call out from a phone in the local network where the asterisk server sits. I turn the debug on, and the log are shown below. I can see "REGISTER" method is OK. ( SIP/2.0 200 OK) But Later, in the "INVITE" method, the SIP addresses become

I'm having a horrible experience getting a Cisco ATA-186 to work with *. I can make calls from the ATA with no problems. However, incoming calls make the ATA ring once, and then the call is disconnected. I have no problems with my Sipura 2000 or my Grandstream phones. I am running 2.16.1 sip code on the ATA 186. Neither * nor the ATA is behind a NAT. They are both on public IP addresses

Hello all, taking examples from various pointers, I am attempting to put together an outbound dialing example using SIP (Cisco 7960) with 2 X100P. Everything seems to be working without generating errors, but the problem is the phone hangs up (102/Bye). Any pointers/advice are much appreciated Here is the section in extensions.conf: extensions.conf ; From CISCO at work ; exten =>

Hi All I seem to have a small problem with the music on hold button on SJPhone. I have 2 asterisk installations one from the Rapid distribution and one from the latest CVS. On the rapid dist when I press the music on hold button on my SJPhone I get music on hold. When I do the same I get no music on hold just silence. I create extension like this exten => 1111,1,MusicOnHold(Default),

I have asterisk with following users; a) zaphfc ISDN card with two channels b) two mediatrix FXS gateways with four channels each c) 1x CISCO 7905G d) two notebooks with MS Messenger 4.7 Now, it seems that any combination works correctly in all combinations except when I call from MS messenger and then call is dropped always in 25th second of the call. Any ideas what I did wrong? here is my

SDP question if * recieves "a=rtpmap:103 telephone-event/8000" it shouldn't it send out the same "a=rtpmap:103 telephone-event/8000" to the other side of the connection? and not something like "a=rtpmap:101 telephone-event/8000"? Thanks

Hello, After I have re-read the "PJSIP Advanced Codec negotiation" document, it occurred to me that the desired behavior should actually happen automatically, just due to the codec negotiation logic, but it looks like asterisk doesn't actually follow the described logic which is likely a bug. Can you please follow with me through a simple sip call and see if I'm missing

Using DIAX softphone which seems to be working OK can get to VM/echotest etc in the demo context Am trying to setup FWD but get the following problems Can hear it ringing when dialing FWD no 612 for time. Connects but no sound from remote end. Does anyone have any suggestions. Softphone on 192.168.0.2 asterisk on 192.168.0.3 Netgear RP114 doing NAT to the internet port 5060 being forwarded to

I am experiencing a problem that from list archive it appears others are running into. When I dial from Cisco 7960 via the * to Free World Dialup destinations that supports G.729 the call fails. The major error from the debug log is Jan 15 00:11:14 NOTICE[22545]: channel.c:1481 ast_set_read_format: Unable to find a path from G729A to ULAW Jan 15 00:11:14 NOTICE[22545]: channel.c:1451

Can someone help me what went wrong with this call? This call was initiated from dev/ttyI0 device on my asterisk server to mediatrix unit. Mediatrix unit user received the call and call started. I can hear them OK but they can not hear me correctly (cut-off sound, noise). Call was finally hunged up. Can anyone point out if there was something wrong? -*CLI> sip debug SIP Debugging Enabled