search for: 2203

...riority 1 in macro): (normal call setup and progress not shown - I show everything after hangup) *CLI> DEBUG[30737]: File rtp.c, Line 739 (ast_rtp_raw_write): Difference is 10888, ms is 1381 DEBUG[30737]: File channel.c, Line 2015 (ast_channel_bridge): Didn't get a frame from channel: SIP/2203-751b DEBUG[30737]: File channel.c, Line 2083 (ast_channel_bridge): Bridge stops bridging channels SIP/2203-751b and SIP/2205-12f1 == Spawn extension (intern-post, 2205, 1) exited non-zero on 'SIP/2203-751b' -- Executing Macro("SIP/2203-751b", "record-cleanup") in n...

...tphone on 192.168.0.2 asterisk on 192.168.0.3 Netgear RP114 doing NAT to the internet port 5060 being forwarded to the asterisk box. This seems to be quite useful software but it's frustratingly difficult to get running. Jon SIP debug shows following mrpenguin*CLI> -- Registered '2203' (AUTHENTICATED) at 192.168.0.2:5036 -- Accepting AUTHENTICATED call from 192.168.0.2, requested format = 2, actu al format = 2 -- Executing SetCallerID("IAX[2203@2203]/9", "91184") in new stack -- Executing SetCIDName("IAX[2203@2203]/9", "calisto&...

...p("This part of the code should never run!") exten => 2231,n,Set(counter=$[${counter}+1]) exten => 2231,n,EndWhile exten => 2231,n,NoOp("This part of the code should be the only thing that gets run!") Console output from dialing 2231: -- Executing Set("SIP/2203-c134", "staticnumber=0") in new stack -- Executing Set("SIP/2203-c134", "counter=1") in new stack -- Executing While("SIP/2203-c134", "0") in new stack -- Executing NoOp("SIP/2203-c134", ""This part of the cod...

Can anyone please help, I can't tell why it will not connect. I do not know how to read this debug file to were it is wrong. Thanks Sip read: REGISTER sip:192.168.1.101 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.100:9082 From: <sip:2203@192.168.1.101>;tag=97442d5b-75b7-4e23-9021-b8605797eb56 To: <sip:2203@192.168.1.101> Call-ID: ea352d6f-a879-4db6-a361-365487a20d4a@192.168.1.100 CSeq: 1 REGISTER Contact: <sip:192.168.1.100:9082>;methods="INVITE, MESSAGE, INFO, SUBSCRIBE, OPTIONS, BYE, CANCEL, NOTIFY, ACK"...

Hi all, I have the below extension defined under sip.conf: [2203] type=friend username=2203 secret=123456 host=192.168.0.164 mailbox=2203 context=intern canreinvite=yes dtmfmode=rfc2833 When trying to register from a softphone installed on a PC behind a nat with IP=192.168.0.164, I got 503 FOrbidden...Does anyone have any idea about what could be the issue? Re...

...runs:0 frame:0 TX packets:5744 errors:0 dropped:0 overruns:0 carrier:0 collisions:0 txqueuelen:0 RX bytes:591811 (577.9 KiB) TX bytes:591811 (577.9 KiB) And this, in /var/log/messages [root at sunspot ray]# tail /var/log/messages Sep 7 06:50:31 sunspot avahi-daemon[2203]: Registering new address record for xx.xx.xx.15 on eth0. Sep 7 06:50:31 sunspot avahi-daemon[2203]: Registering new address record for xx.xx.xx.20 on eth0. Sep 7 06:50:31 sunspot avahi-daemon[2203]: Withdrawing address record for xx.xx.xx.20 on eth0. Sep 7 06:50:31 sunspot avahi-daemon[2203]: R...

https://bugzilla.mindrot.org/show_bug.cgi?id=2203 Bug ID: 2203 Summary: scp hangs with -f option Product: Portable OpenSSH Version: -current Hardware: All OS: Linux Status: NEW Severity: minor Priority: P5 Component: scp Assig...

Hi there, do I have a local problem, or is registration at IAXTEL impossible at the moment? "iax2 show registry" permanently shows a TIMEOUT for 69.73.19.178. Philipp

Hi all, New user to asterisk having just got it compiled and installed. Running with no digium hardware (yet) and no soundcard in asterisk box. Problem is using the sample configs with a sip phone added as follows [2203] type=friend username=2203 secret=2203 host=dynamic defaultip=192.168.0.2 dtmfmode=inband canreinvite=yes the console on * when running with -vvvvc says :- (whenb trying to dial extension 500 (the demo) from my xten sipphone) -- Executing Playback("SIP/2203-2e99", "demo-abouttotry&...

Hi there, I have tried very hard to setup the x-lite with asterisk, but until now i didn't get sucess. When i start the asterisk in debug mode, i see the message: sip/2.0 401 unauthorized. I know that this problem with authentication. I put in my sip.conf as below. [2203] type=friend username=2203 auth=md5 secret=1234 reinvite=no canreinvite=no dissallow=all allow=gsm context= sip host= 192.168.10.149 -> my machine that have xlite extension.conf [sip] exten => 2203,1,Dial(${Phone1}) I have read and read many message in list but i could found an...

Good Day again list, Encountered another problem in the ACD queue... If I use the ADDQueueMember to dynamically add members as foolows, exten => 403,1,AddQueueMember(techsupport|SIP/${CALLERIDNUM}) lets assume I called extension 403 from my extension 2204. then a caller (extension 2203) enters into the techsupport queue I am able to receive the support call on my phone (extension 2204 rings). I take the call and am talking with the customer (extension 2203). Now if extension 2201 calls into the techsupport queue, The problems rears its ugly head. One would think that the...

...ed is a list of the files I think you may need to look at. Agents.conf, extensions.conf, queue.conf, and the CLI log output. Thanks in advance ~ron Log files: <************* Had someone dial 9999 to go into the Techsupport Queue ********************> -- Executing Queue("SIP/2203-71a6", "techsupport|t||tech_que|944") in new stack -- Started music on hold, class 'default', on SIP/2203-71a6 -- Stopped music on hold on SIP/2203-71a6 -- Playing 'queue-youarenext' (language 'en') -- Told SIP/2203-71a6 in techsupport their qu...

...EST}) exten => 2209,3,NoOp(${OTHERTEST}) exten => 2209,4,Dial(SIP/2209) exten => 2209,105,Answer exten => 2209,106,Playback(invalid) exten => 2209,107,NoOp(${MYTEST}) exten => 2209,108,NoOp(${OTHERTEST}) exten => 2209,109,Hangup ms1*CLI> -- Executing SetVar("SIP/2203-2496", "MYTEST=ishouldnotseethis") in new stack -- Executing Dial("SIP/2203-2496", "Local/2209@local") in new stack -- Called 2209@local -- Executing SetVar("Local/2209@local-2af6,2", "OTHERTEST=goodness") in new stack --...

...ocancelwhenbridged=yes channel => 1 ; ; FXS Port 1 context=local signalling=fxs_ls usecallerid=yes echocancel=yes echocancelwhenbridged=yes ; ;FXS Port 2 context=local signalling=fxs_ls usecallerid=yes echocancel=yes echocancelwhenbridged=yes /etc/asterisk/extensions.conf [local] exten => 2203,1,SetMusicOnHold,loud exten => 2203,2,Dial(Zap/2,15,Ttr) exten => 2203,102,Voicemail(2203) exten => 2203,Hangup exten => 2204,1,SetMusicOnHold,loud exten => 2204,2,Dial(Zap/2,15,Ttr) exten => 2204,102,Voicemail(2203) exten => 2204,Hangup Should this be enough for me to get...

Can asterisk act as a SIP registrar or location server? I would like to be able for a user agent(client) to register with whatever client they are using as "username@domain-name.com". Rather than the entry/username/password that is setup in the sip.conf file. That way a user could log into any SIP enable client and their calls would follow them around. I have read the sip.conf man pages

...same US numbers with the service by using a direct connection from a softphone for example. The entries that show up in the log after failed attempts to call the US are: Jul 11 20:04:04 WARNING[25225728]: File channel.c, Line 1851 (ast_channel_make_compatible): No path to translate from SIP/2203-2929 (4) to IAX2[vbx]/1(16) Jul 11 20:04:04 WARNING[25225728]: File app_dial.c, Line 672 (dial_exec): Had to drop call because I couldn't make SIP/2203-2929 compatible with IAX2[vbx]/1 I don't see anything suspicious entries in the CLI logging with IAX2 debugging on. Searching the...

...cpe ; p2mp NT mode ;signalling = bri_net_ptmp ; p2p NT mode ;signalling = bri_net pridialplan=local prilocaldialplan=local echocancel=yes immediate=yes group = 1 context=demo channel => 1-2 Zaptel.conf: loadzone=nl defaultzone=nl span=1,1,3,ccs,ami bchan=1-2 dchan=3 Example where a sip client (2203) is calling xxxx7024

...ded to look at something else and looked at the box a couple of hours later. The problem now didn''t occur anymore. I could execute all commands without hanging. This log file shows errors from the snmpd daemon. Every 30 secs, a repetition of the three lines: Nov 6 09:41:45 alpha snmpd[2203]: netsnmp_assert index == tmp failed if-mib/data_access/i <<snip>> Nov 6 09:41:45 alpha snmpd[2203]: netsnmp_assert __extension__ ({ size_t __s1_len, __s2_l <<snip>> Nov 6 09:41:45 alpha snmpd[2203]: err...

...redentials. Why this happen? When I call from the outside my number at addaline, asterisk answer and send me to my soft phone without any problem. My settup is something like this: sip.conf [proxy.addaline.com] type=friend host=216.87.144.203 username=5555555555 secret=secret context=from-sip [2203] type=friend username=2203 secret=secret host=xx.xx.xx.xx mailbox=2203 context=intern canreinvite=yes dtmfmode=rfc2833 extension.conf: [from-sip] exten => _.,1,Answer exten => _,1,Dial(SIP/2203,20) exten => _,2,Hangup exten => 2203,2,Voicemail(u2203) [intern] exten => _7.,1,Dial,S...

Does anyone in Asterisk land know how to turn on the message light on the back of the earpiece of a cisco 7960 when a message is waiting? Thanks! Paul Paul Mahler mail:pmahler@signate.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040130/0efacc79/attachment.htm