similar to: Queues, SIP channel and "In Use"

[Mar 20 12:45:33] WARNING[4940]: app_queue.c:3136 try_calling: The device state of this queue member, SIP/3617001000, is still 'Not in Use' when it probably should not be! Please check UPGRADE.txt for correct configuration settings. [Cary Fitch] We are running 1.4.22 and this message popped up in console. It could be causing our Queues announcement problem, because if all members

Or, in full: [Sep 20 17:11:26] WARNING[18373]: app_queue.c:2705 try_calling: The device state of this queue member, SIP/612, is still 'Not in Use' when it probably should not be! Please check UPGRADE.txt for correct configuration settings. So, what do I check in UPGRADE.txt? This is with Asterisk 1.4.11

Hello list, asterisk 1.4.30 2 situations in which call-limit should work, but it does not : [Jul 8 09:15:49] WARNING[11132]: app_queue.c:3272 try_calling: The device state of this queue member, test12, is still 'Not in Use' when it probably should not be! Please check UPGRADE.txt for correct configuration settings. In sip.conf I have : limitonpeer = yes In my realtime sip_buddies

HI Im getting strange message on asterisk console WARNING[26853]: app_queue.c:2321 try_calling: Announcement file 'custom/announce-adslsetupnatrate' is unavailable, continuing anyway... thanks arun -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070603/6564c117/attachment.htm

hello all, I have asterisk 1.4.26 working on CentOS 5.3 and I have the following problem: - when I restart asterisk all the members of the queue are Invalid. - when I make a call to one of the members, of the queue, and then check the state, it turns to "Not in use" for the called phone, and the queue works fine for that member after. - after doing a module reload of the

Hi, I'm wondering - why SIP device state doesn't get updated to anything else, except Not In Use. For queue call (with Local channel) i get: app_queue.c: Device 'SIP/21168' changed to state '1' (Not in use) app_queue.c: Device 'SIP/21168' changed to state '1' (Not in use) app_queue.c: The device state of this queue member, Agent/21168, is still 'Not in

When working with call queues, if an agent is logged in via AgentCallbackLogin and the extension they are registered at becomes "unavailable" (from a bad connection, or something of the like), calls routed to that extension seemed to be accepted by it, so if the next action for that extension is to go to voicemail, the caller in the queue is sent to the extensions voicemail. Even worse,

Hi, I'm using queue configuration as follows: - queues from* queues.conf* - queue_members from *external Database thru ODBC*, using* Local channels * as interface - sip extensions from *external Database thru ODBC* When a call is sent from queue to an interface (local channel), it is answered but a message appears at the CLI: *[Nov 4 16:56:04] WARNING[13951]: app_queue.c:3014

2023-11-08 18:14:13] ERROR[571246][C-000017e2] : Got 19 backtrace records # 0: [0x5bd18a] asterisk utils.c:2800 __ast_assert_failed() # 1: [0x4618e3] asterisk astobj2.c:589 __ao2_ref() # 2: [0x58e660] asterisk stasis_cache.c:824 update_create() # 3: [0x58efed] asterisk stasis_cache.c:903 caching_topic_exec() # 4: [0x586b90] asterisk stasis.c:1380 dispatch_message() # 5: [inlined] asterisk

Hello, We're using Asterisk to manage call queues. Queue members are connected via IAX2 using the Zoiper softphone, and Zoiper is configured with 2 lines. We're finding that calls are routed to queue members even when they are on the phone, on their softphone's other line. For example, if a queue member makes an outgoing call on line 1 or is handling a queue call on line 1, the

Hi, I'm using the following setup : Alice ---- IPPhone ------<LAN>----- Media gateway ----<PSTN> ------- Phone ---- Bob For certain calls, users complains about echo : they can ear their own voice in their handset, though media gateway echo cancel is turned on. I'm wondering how this echo cancelation engine is supposed to work. My understanding of echo is that most probably,

On Tue, Dec 23, 2014 at 6:51 PM, Joseph <syscon780 at gmail.com> wrote: > > > Most cell phone don't have a USB port but you are correct, maybe I just need > IAX2 soft-phone like: > Zoiper - it works on most of the platforms. I think Zoiper registers > directly with Asterisk IAX2 (if configured) as an extension, isn't it? If your cellphone is capable of a Wi-Fi

Greetings List. we're facing a strange case with my system where in the middle of the call .. after like 7 minutes (not necessarily ) the callee is unable to hear the caller however the caller is able to hear the called party. the scenario is the following. 1- 15 computers running Windows XP SP3 joining a Windows Domain Controller with DHCP , DNS, ISA Internet Acceleration Server. 2- Internet

Hello I need to hook up someone's remote PC onto our Asterisk server over the Net. There are firewalls on each side, so I figured it's time to give IAX a try, and see if it's less of a pain to use than SIP. And since IAX hardphones are pretty are, I guess I'll go softphone. Apparently, the two most well-known IAX and SIP clients for Windows are ZoIPer and X-Lite, respectively.

How can I set the prefered codec between 2 calling parties ?? My Grandstream supports G729, alaw and gsm... in this order. The Zoiper softphone has alaw and gsm as codecs... in that order. Although there should be a matching codec found, my Grandstream can not call the Zoiper softphone. CLI shows : [Mar 11 17:47:21] WARNING[22367]: channel.c:3340 ast_channel_make_compatible: No path to

Hello. I've been trying to use the AMI to originate phone calls. I'm trying to call the SIP phone 'zoiper' with the SIP phone 'yziquel'. So, the AMI interaction is: > Action: originate > Channel: SIP/zoiper > Exten: yziquel > Priority: 1 > Timeout: 30 > Context: internal > > Response: Error > Message: Originate failed > > Event:

I'm trying to setup zoiper ( formerly idefisk ) to use my asterisk server at work from home. I've setup zoiper for iax, set the ip address to work's fixed ip address, user: home, password: password but the zoiper log shows: 11:02:35 Rejected registration for 'home@<my-office-ip-address>' with cause 'facility rejected' 11:03:35 Rejected registration for

Hello guys, The so expected 2.0 release of Idefisk 2.0 softphone is a fact. Idefisk and Zoiper became one - Zoiper 2.06. Here are some of the features: SIP and IAX, TCP, TLS support, Multi-language support, Automatic provisioning (XML), URL handling, Outlook Integration, Native conferencing, API, Changeable number of lines.... You could read the complete Press Release here:

Hi all, I've setup two * servers which are SIP interconnected ala osaka/toronto from the * book (before anyone sugggests using IAX instead, no, I NEED to have them SIP interconnected for verification/test purposes). Then I have a Zoiper connected to one of them via IAX (so that * will not reinvite (?)). As soon as I try to call (via Zoiper) an extension on the other * I get a "Failed to

I used to use IDEFISK, but since it was taken over/renamed into Zoiper it's been really hard work - now I'm told that they won't support my chosen distribution - Debian Etch - the current stable version of Debian I prefer. So, looking to dump Zoiper and go with something else - I want something light-weigh (So that rules out Ekiga - and Zoiper was going down the bloatware route