similar to: Red: Sip Phone CID

Hello, Has anybody seen that Audiocodes gateway is replying with "486 Busy here" when it's actually not (last call ended ~15 seconds ago). I see this quite often. From other logs i see that previous call ends at 11:13:01, then app_queue tries to dial at 11:13:14 and fails numerous times, before succeeding at 11:14:02 I have attached sample SIP debug log: Any ideas what i could

The following transfer led to a crash of asterisk, without leaving a core or any utterances in messages or debug file. It looks like the zombie which was created during the MASQ-transfer was not cleaned up... But why did it start a Dial??? And... why does Asterisk die when this happens?? Thanks!!! Michiel -- Zap/32-1 answered Zap/6-1 -- Stopped music on hold on Zap/6-1 -- Starting

hello, I have a test setup with 2 asterisk servers, each having a one snom 100 via sip using it. I`m experimenting on how trunking between them would work. I have them setup for RSA authentication which I plan to use in the future. So I`ve setup the keys and servers seem authenticate to each other. One is named phila and other hurricane. Here is what I see on phila: -- Registered

Hi, In this thread http://lists.digium.com/pipermail/asterisk-users/2008-October/219592.html , I wondered whether SIPPEER curcalls was working. I could test this anew today. Here are my findings : Alice, Bob and Carol ar all using SIP Phones. Whenever Alice is calling Bob, - if Carol is calling Alice, SIPPEER(Alice:curcalls) equals 0 - if Carol is calling Bob, SIPPEER(Bob:curcalls) equals 1

Or perhaps how to configure and refer to two parallel IAX trunks with different codecs? I have a situation where I'm using G.729A as my IAX trunking codec. Now I need to push some short duration, low bitrate modem traffic over the link (a credit card terminal). Obviously the modem audio isn't going to survive the G.729 codec process intact, so for the times the device is used I'd like

On Wed, Jan 20, 2010 at 2:42 PM, JR Richardson <jmr.richardson at gmail.com> wrote: > Hi All, > > I'm using Asterisk 1.4 branch and checking the status of some SIP > Peers with the functions ${SIPPEER(101:status)} and the result is "OK > (48 ms)". ?Seems to work fine. > > Now I would like to use the function CUT to set a variable with the > 'OK'

Hello all, I'm struck with a very strange problem today. I've an AGI with some code subroutine snippet as follows: sub enable_sbc($) { my $carrier = shift; my $tmp = substr($carrier,1); my $jkh = $tmp; $server_port = $ast_agi->get_variable("SIPPEER($jkh,port)"); $ser_ip = $ast_agi->get_variable("SIPPEER($tmp,ip)");

Hi, i have the same problem. Trying to use the dialplan function CONNECTEDLINE() this way Set(CONNECTEDLINE(name)=${SIPPEER(${EXTEN},callerid_name)}) Set(CONNECTEDLINE(num)=${EXTEN}) ends with [Jul 1 16:30:50] ERROR[3954]: pbx.c:2902 ast_func_write: Function CONNECTEDLINE not registered Same happens trying function CALLEDID. I am using Asterisk 1.6.1.20. What do i

I'm simply trying to forward calls to users who have the call forwarding feature enabled (FWD Status and FWD Ph Number kept in the astDB). The problem is that I want users to be able to forward calls to numbers that they would normally be allowed to dial within their own context. (I don't want a local call only person forwarding to a long dist number, for example.) I'm able to

try, database get SIP/Registry/<peername> it gives you a string which contains the info, then pass it to CUT to extract ip-adr and port Freddi > To do that you need to get the remote ip address and port of the sip peer! > > I found the function: > > ${SIPPEER(exten:ip) > > But how can I get the port??? > >

On 1.6.2.11-rc2 I've noticed a bunch of DEBUG statements on startup, such as: == Registered custom function 'SIP_HEADER' [Aug 9 07:01:23] DEBUG[17330]: xmldoc.c:1727 xmldoc_build_field: Cannot find variable 'SIPPEER' in tree 'description' == Registered custom function 'SIPPEER' [Aug 9 07:01:23] DEBUG[17330]: xmldoc.c:1727 xmldoc_build_field: Cannot find

Howdy, I'm trying to re-write my voicemail check extension. I formerly used the SIPPEER function to get the mailbox for a peer with ${SIPPEER(${peer},mailbox)} Is there a way to do this with PJSIP now that I've converted over? I see a function PJSIP_ENDPOINT and it has a mailboxes subset but I'm not retrieving any data from it when I query it. -- A human being should be

I'm still struggling to pickup calls. I now have a single context (entryocginternal) where I have "include => parkedcalls". The log below shows me calling from one internal extension to another, then picking up, then parking the call. -- SIP/239-0915d5c8 is ringing -- SIP/239-0915d5c8 answered SIP/233-0915bf40 -- Packet2Packet bridging SIP/233-0915bf40 and

Hi All, I'm using Asterisk 1.4 branch and checking the status of some SIP Peers with the functions ${SIPPEER(101:status)} and the result is "OK (48 ms)". Seems to work fine. Now I would like to use the function CUT to set a variable with the 'OK' portion of the status "OK (48 ms)" and then do some follow on stuff if the status is OK. I'm running into syntax

This code executes just fine, and leaves the SIP peer's mailbox setting from sip.conf in variable target. exten => 1,1,Set(target=${CHANNEL:4}-) exten => 1,n,Set(target=${SIPPEER(${CUT(target,,1)}:mailbox}) exten => 1,n,VoiceMailMain(${target}) However, every time it runs I get an error in the CLI as follows WARNING[5629]: pbx.c:1366 ast_func_read: Can't

Hi, In sip.conf, it's possible to add a line such as setvar=MYFIELD=foo and access this value from diaplan with SIPPEER function. 1. Which function is available to access values in users.conf such as vmsecret ? 2. Is it possible to extend users.conf with custom keys/values ? Regards. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I know you can access various sip variables via 'Set(sstatus=${SIPPEER(201,status)})' (for example) to get the status of the sip user - but what about variables? I have a user that has setvar=123456 in their users.conf (sip.conf if you prefer). I can read it with a 'sip show peer 201' - but that gives everything and parsing that isn't really an option. Anyone know how

Hi gang, Since I'm getting no joy from device_Status or SIPPEER in 1.4.26-rc1, I thought I would do an AGI to read my hints and check for line in use that way. The AGI works fine from a prompt, but returns the dreaded "utils.c:966 ast_carefulwrite: write() returned error: Broken pipe" when I try to run it from the dialplan. Here is my dialplan snippet;

Ok, i tried the suggestion: Instead of: sippuser => resource, database_name, table_name sippeer => resource, database_name, table_name I put in: sippuser => resource, context, table_name sippeer => resource, context, table_name Unfortunately, with the same results. btw i tried both "general" as "default" Besids the commands i tried below, isn't there any

2008/11/4 Igor Zamocky <asterix at ponozky.sk> > > Did You tried http://www.voip-info.org/wiki/view/Asterisk+sip+limitonpeers > ? > I didn't. Now I did and it's working the way I wanted. Meanwhile, I had found a (complex) workaround using GROUP, GROUP_COUNT and SIPPEER but limitonpeers is much more concise. Thanks a lot. > > > Hi, > > > In this