similar to: asterisk 10.4.2 realtime callgroup missing

Hi, I am using Realtime for sip configuration. When there is an INVITE which arrives at asterisk asterisk makes the following selects: Nov 20 10:37:10] DEBUG[31852]: res_config_mysql.c:651 mysql_reconnect: MySQL RealTime: Everything is fine. [Nov 20 10:37:10] DEBUG[31852]: res_config_mysql.c:138 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM sip_conf WHERE name = 'tzl' [Nov

Hi, I asked a time ago what were callgroup and pickup group used for. I have done some proofs and all, and I'm not sure if I have pick the idea up well!! That's what I understand: For example: group=1 callgroup =2 and pickupgroup=2 and my phone is a membership of the group 1. that's mean that when a phone that belong to group 2 is ringing, I'll be able to answer this call dialing

I'm trying to set up asterisk so any phone connected to channel 1-16 of my Adit600 channel bank can pick up a call coming in on channel 24. I do not wish to ring any of the 16 channels on an incoming call -- this is strictly so I can pick up the line if I see it ringing and wish to answer at work. I have channel 24 in call group 3, and channels 1-16 in pickup groups 1 and 3. However

I'm trying to get sipusers working with a realtime odbc database on Asterisk 1.6. We have sippeers working from the database, but need sipusers to be in a separate table for other implementation reasons. sip show user test load returns results from the database. CLI> sip show user test load * Name : test Secret : <Set> MD5Secret : <Not set>

Hi After recently upgrading to 1.8.3 I have noticed that the nat setting for my peer in my sip table is not making it into the realtime cache. For example * Name : 501 Realtime peer: Yes, cached Secret : <Set> MD5Secret : <Not set> Remote Secret: <Not set> Context : pack-local Subscr.Cont. : <Not set> Language : AMA flags :

HI,ALL, I have multiple PSTN lines registered as multiple SIP channels (e.g. SIP/line1, SIP/line2, SIP/line3, etc...), on the multiple gateways( I uses the SIPURA3000). I wants to arrange them into an ordered hunting group for outbound calling. I used http://www.voip-info.org/wiki/view/hunt-dial+macro for reference. My configure files like blew.

Hello, I installed 1.2.5 and realtime SIP. The connection to the DB is OK because I can get the values from the CLI. Here are my 3 different cases: 1- If I put an unexisting user, I get 404 and I am not able to dial. 2- If I check "Disable registration" within Firefly it does not register but I am able to dial a destination (...) 3- If I leave registration ON, I get the 404 message

Hi Gurus. I searched the lists, wiki and the rest of the web but I still do not understand this. My Setup is as follows: [ISDN via chan_capi or IAX2 DiD Provider] => [* PBX] => [IAX2 Clients (Atcom AT-320ED)] I want to get callgroup/pickupgroup and callwaiting working on the IAX phones. Some web sources told me that this was not implemented, others say that the phone has to handle

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`

Hello Asterisk-Community, I'm having an error with my BLF configuration on my asterisk...i've configured the sip peer like this: [8250] type=friend callerid=Extensi?n 8250 <8250> canreinvite=no context=pbx9 dtmfmode=rfc2833 host=dynamic insecure=no language=es nat=yes pickupgroup= callgroup= qualify=2000 secret=cyx2mo type=friend username=8250 subscribecontext=pbx9 call-limit=100

I have configured asterisk realtime to work with two servers and a seperate MySQL DB. Each sip client registers which server it is connected to in the MySQL DB. This works great as long as the clients are 1. On the same network 2. Behind a NAT and connected to the same asterisk server as the caller. However I need this configuration to work for "NAT-ed" clients on different asterisk

Hello, Right now there is not a table build script at: http://www.voip-info.org/wiki-Asterisk+RealTime+IAX Therefore I have taken the SIP build script and added a few fields that I use from my iax.conf (could be more out there, please see the complete build script below): `dbsecret` varchar(100) default '', `notransfer` varchar(100) default '', `inkeys` varchar(100)

Hi All, I am trying to configure asterisk realtime. But i am unable to get the extensions listed successfully when i type "sip show peers" in the asterisk CLI . i am unable to see any failure logs when i do a reload i can able to connect to the data source through "odbc show" in the CLI, Any hep in this regard is highly appreciated. Following is the configuration

I have problems using the call pickup under Asterisk 1.8.4.2. I have another Asterisk with 1.6 - and it is working fine with the same settings. I have setup the same callgroup and pickupgroup for all extensions in sip.conf - just to make things simple for testing. The sequence *8 seems to be completely ignored by Asterisk - the client shows "Call answered" when dialing *8 while the

Trying to get SIP realtime working here... I'm connected to the database... *CLI> realtime mysql status Connected to vox180internal@db1.ipt.XXX.com, port 3306 with username voxadmin for 6 seconds. I can get information for the extension in question... *CLI> realtime load sipusers name 2944093 Column Name Column Value

Hi all, at time i am trying to get a better idea of callgroups and pickupgroups (especially within the SIP Channel) A Pickupgroup is relative clear - everyone in the same pickupgroup may pickup a call And a callgroup does what ? - The same ? I thought that a callgroup would act like the ZAP groups - so that you then can dial SIP/g1 - and every SIP Client which is in the callgroup 1 does then

I have just started to play with callgroups and pickupgroups.. I updates my * from CVS this morning (about 15 mins ago).. I have placed callgroup=1 and pickupgroup=1 into each of my 3 phone configurations in sip.conf.. I place a call from phoneA to phoneB, then I go to phoneC and dial *8# , the call does not get picked up by phoneC and continues to ring on phoneB.. Have I not configured

Am I reading the data below incorrectly, or does it appear that even though I have the directive canreinvite=no set for the two asterisk boxes, they are trying to do a reinvite (which fails) anyway? Is this expected behaviour in this situation? If so, how can I prevent this? ---- Lots of output ---- Using CVS Head from 2005-08-28, I have two asterisk boxen, one (box A) has a sip ua (2608)

Suddenly the other day we noticed MWI stopped working for SIP clients. A "sip show peer X" returns this: ast01*CLI> sip show peer 719XXXXXXX * Name : 719XXXXXXX Realtime peer: Yes, cached Secret : <Set> MD5Secret : <Not set> Remote Secret: <Not set> Context : peakinternet-outbound Subscr.Cont. : <Not set> Language :

Callgroups/pickupgroups are allocated per channel, not in the dialplan. sip.conf and zapata.conf are the two files you're interested in. -wade ---- Original Message ---- From: wipeout@linuxmail.org To: asterisk-users@lists.digium.com, Subject: RE: [Asterisk-Users] Using callgroups (was: Taking a call for someone elses extensionfrom my extension) Date: Sun, 20 Apr 2003 16:39:15 +0000