similar to: IAX -> IAX -> SIP problems

This just started happening today. I've got 1 queue and 6 agents. All logged in. I tell the service people to ignore my call if they see my caller id. I call the queue and watch as asterisk bounces me around the phones. Our agent ring time is 5 second timeout and a 5 second wait time before trying next agent. I get the same message in console for each agent attempt: -- Executing

I'm not a PRI expert and therefore don't know what this debug stuff means for PRI, so if anyone can help me here... I'm running the latest libpri and zaptel from CVS. Keep in mind that everything works fine when using the STABLE libpri and zaptel. I am NOT running CVS asterisk. I am running 1.0.6. asterisk*CLI> pri intense debug span 1 Enabled EXTENSIVE debugging on span 1 T203

Any ideas on this warning? If I call this number, sometimes I get this error and sometimes the call goes thru fine. Why would it work sometimes? -- Executing Goto("SIP/3044-8d49", "cytel-outgoing|915124512424|1") in new stack -- Goto (cytel-outgoing,915124512424,1) -- Executing SetCIDNum("SIP/3044-8d49", "2814494000") in new stack --

Hi, I have a dataframe, hilodata, which looks like this: > hilodata sym date maxprice minprice ntick 1 ABK 19910711 11.1867461 0.0000000 108 2 ABK 19910712 11.5298979 11.1867461 111 3 ABK 19910715 11.7357889 11.4612675 52 4 ABK 19910716 11.5298979 11.3240068 51 5

Another example: Jerry Friedman's MART is available in R from Salford for the same price as the stand-alone TreeNet, even though they don't advertise it on their web site. Andy > -----Original Message----- > From: rossini at blindglobe.net [mailto:rossini at blindglobe.net] > Sent: Sunday, August 17, 2003 9:50 PM > To: rhelp > Cc: pralay at cytel.com > Subject: [R]

At JSM, I spent a bit of time with old friends at the Cytel booth (makers of StatXact/LogXact). They were wondering whether it was both feasible and of interest to create a package of the StatXact compute engine for R (to be commercially licensed, not for free!), similar to what they've done for SAS. As far as I know, it's feasible, (this is not the first commercial external package,

Why are there so many attempts to native bridge? The call is actually up and working by attempt #1 so what is it doing on all those other attempts? We are only allowing G729 so it can't be codec negotiation. -Matthew -- Executing Goto("SIP/3013-3dfa", "cytel-outgoing|917034439032|1") in new stack -- Goto (cytel-outgoing,917034439032,1) -- Executing

This is a call coming in through the ISDN to 7940's. Answering with non-codec capability 1 - Is that the problem? SIP Debugging Enabled We're at 10.1.0.11 port 18406 Answering/Requesting with root capability 8 Answering/Requesting with preferred capability 4 Answering/Requesting with preferred capability 8 Answering with non-codec capability 1 <<<<<<------------- 12

Hi all I have a problem to register a cisco 7960 to an asterisk 1.2.2 I defined in sip.conf the next : ["phonenumber"] type=friend username="username" secret="password" host=dynamic context=work I am trying to catch the register requests with sip debug with no success (empty screen). I can only catch the register messages with ngrep on

Hi all I have a problem to register a cisco 7960 to an asterisk 1.2.2 I defined in sip.conf the next : ["phonenumber"] type=friend username="username" secret="password" host=dynamic context=work I am trying to catch the register requests with sip debug with no success (empty screen). I can only catch the register messages with ngrep on host it's comming

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

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'm using the Debian Sarge package of Asterisk - 1.0.7 + bristuff. When I call from a zap channel or a SIP phone to another SIP phone, then dial *8 from a third SIP phone, I get 503 Service Unavailable on the third phone and I get this at the Asterisk console: Jun 28 09:01:24 DEBUG[16774]: res_features.c:1709 ast_pickup_call: No call pickup possible... Jun 28 09:01:24 NOTICE[16774]:

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

I've been banging my head on this for several hours, and I have no idea what's going on. Maybe there is a very simple result, and I've been looking too hard at this this evening. This is a brand new system, and I'm wondering if there have been SIP bugs introduced in the latest CVS that are preventing from working what should be a stupendously simple test. - Cisco 7960

Hi everybody, I've got a very strange problem: As far as I remember I didn't change anything on my Asterisk side. I have 2 SIP providers to which I can place outbound calls. Today I noticed that outbound calls to provider "inode" fail (and inbound from this provider too). On the CLI I get every 20 seconds following messages: Nov 9 20:01:07 NOTICE[952]: chan_sip.c:5422

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 =>

ok so my table looks like this... REATE TABLE `sip` ( `id` int(11) NOT NULL auto_increment, `name` varchar(80) NOT NULL default '', `accountcode` varchar(20) default NULL, `amaflags` varchar(7) default NULL, `callgroup` varchar(10) default NULL, `callerid` varchar(80) default NULL, `canreinvite` char(3) default 'yes', `context` varchar(80) default NULL, `defaultip`

Dear all i'm trying to originate an outgoing call with the command originate, from Asterisk's CLI i'm typing: CLI> originate IAX2/my-iax-provider/number2call application wait 10 [Apr 18 19:31:12] DEBUG[32331]: chan_iax2.c:4000 create_addr: prepending 40 to prefs -- Call accepted by 62.149.202.150 (format ilbc) -- Format for call is ilbc -- Hungup

Hi, I'm having a problem with the queue behaviour in my place: I have two ISDN channels to the outside (Zap/1) and two channels two a Siemens Gigaset (Zap/4). I also use a SIP gateway to call outside and have a couple of IP phones around as well (SIP). The Gigaset has about 5 phones connected to it (+base station). Whenever two people are using those, I always am blocking two internal