similar to: Asterisk 12 and chan_local

Hello, I'm trying to build Asterisk12 on a Centos 6.4 VM. The configure script is erring out with: ? checking for uuid_generate_random in -luuid... no checking for uuid_generate_random in -le2fs-uuid... no checking for uuid_generate_random... no configure: error: *** uuid support not found (this typically means the uuid development package is missing) I have installed (using yum) uuid, uuidd

Hello, I've recently given a try to a Digium D70 phone. At the moment, I'm configuring them though config files with a DHCP server and not using DPMA. Of course, I'm connecting them to Asteris (PJSIP stack on 13.7.0). Which is the best place to: - read about past issues - open new tickets for remaining issues. Best regards -------------- next part -------------- An HTML attachment

Happened twice. There might also be a race condition and some bad pointers in chan_local.locals_show. First the segfault. CLI> show locals <unowned> -- 6001@default Segmentation fault (core dumped) [root@mars asterisk]# ll -tr total 22260 [...] Loaded symbols for /usr/lib/asterisk/modules/chan_local.so #0 __pthread_mutex_lock (mutex=0x5d8) at mutex.c:99 99 mutex.c: No such file

Hello I just coding a AGI script for billing. - For external calls, I pass the call directly on a trunk. I do : Dial(trunk1/extension) -> OK ! - For internal calls (shortcode, others users ...) I am Dial(Local/extension at context/n) The problem is that through chan_local.so, I sound as it cut! Example if I call the voicemail ... "You have No messa ..." or "You have

Hi all Have recently watched Matt Jordan's session on Kamailio World 2014 On slides 26-29 of his presentation (http://www.kamailio.org/events/2014-KamailioWorld/day1/09-Matt.Jordan-Asterisk12-And-PJSIP.pdf) he speaks about a (completely new, for me at least) approach to build scalable telephony systems, using N instances of Kamailio and N instances of Asterisk Are there any

Hi, I've been getting regular CPU usage spikes(50%-80%), due to asterisk (according to top). I never noticed this on 1.4, and I have top running in the background pretty much all the time. In between those spikes Asterisk stays under 10% CPU usage (I have a transcoder card, which helps). It's very regular, never any missed spike, or any spike in between the regular spikes. I

Hi, I've recently upgraded from Asterisk 1.4.22 to 1.4.29, and I'm now having a problem with Originate and chan_local. I'm using the following Manager API action to originate a call: Action: originate Priority: 1 Context: trunk Callerid: 100 Channel: Local/100 at callback/n Exten: 123456789 Variable: USERFIELD=127.0.0.1|USEREXT=123456789 WaitTime: 30 This is intended to first call

I have a server with an asterisk ss7 link connected to a Siemens working well for over a year. A few days ago I started having problems with signaling. I found the following logs in / var / log / messages Sep 12 11:49:25 call3 kernel: [1018427.030959] dahdi: Master changed to TE2/0/2 Sep 12 11:49:25 call3 kernel: [1018427.120740] dahdi: Master changed to TE2/0/1 Sep 12 11:49:26 call3 kernel:

The TDM410P doesn't support 'hvac', only the obsolete TDM400P supports that option was for the old phones that have a neon light (or equivalent LED+ZENER ciruit). Are other phones off the TDM410P (other than the VTECH) working, or is the Vtech the only model with VMWI available to you. I'm not able to check at the moment, I have copied the asterisk-users list, someone else may

Hi All, I have installed asterisk 1.6.2.8 Dahdi: 2.4.0 Digium card: Digium, Inc. Wildcard AEX800 8-port analog card I have configured this card properly and it is working for calls too, But there is issue of one way audio on this outbound routes only, My voice is going to outside pstn number but i am not able to get their audio, I have disabled firewall, selinux is also off. If I am applying

Does anyone know if the TE122 is recognized by any of the 1.2 zaptel drivers? It seems that 1.2.16 knows it not... :) Cheers, j

I just installed an FXS module onto a 4 channel tdm thats about 5 years old and it wont work. Running dmesg I can see the following error: Zapata Telephony Interface Registered on major 196 Freshmaker version: 71 Freshmaker passed register test !!!!!!! LOOP_CLOSE_TRES iREG 1C = 1 should be 1000 !!!!!!! RING_TRIP_TRES iREG 1D = 8000 should be 3600 !!!!!!! COMMON_MIN_TRES iREG 1E = 0 should

I just installed a TE820 octal span T1 card, and it's not showing up in dahdi_hardware output. This was installed into a test machine that already has a TDM800P card in it, and that one is showing up and working fine. Is there some kernel module that I'm missing? Lspci: 05:04.0 Ethernet controller: Digium, Inc. Wildcard TDM800P 8-port analog card (rev 11) 21:08.0 Communication

Hi, I'm reading Digium HA8 manual (page 55). It lists several parameters that can be used to configure a BRI port. More precisely, it suggests that : Term, is the last parameter NT/TE is the one just before (starting from the end). If I'm not mistaken, it seems dahdi_genconf generates /etc/dahdi/system.conf files in which both parameters order is reversed : # Span 1: WCBRI/0/0

I have been doing a lot of reading forums and elsewhere but am somehow unable to connect the dots. Here is what I am trying to accomplish initially and then wish for it to grow bigger from here on. I have two POTS (Analog) line that would connect to the Asterisk Box. I have, to begin with 5 IP phones (PoE), all connected to a switch. Asterisk Box with a LAN card also connects to the same switch.

I am having difficulties with customer-bound DTMF being very short & clipped off (and basically unusable, as systems on the customer side aren't recognizing the DTMF digits, and I can barely tell that DTMF is there when I listen on a handset). My system set up as follows: PSTN <--> Metaswitch <-SIP-> Asterisk <-SIP or IAX2-> CPE Asterisk is running Asterisk 10.4.0 on a

?Asterisk 13.11 rc1 ./configure LDFLAGS="-z muldefs" --libdir=/usr/lib64 --with-unixodbc=$(odbc_config --include-prefix)/ --disable-dev-mode --with-pjproject-bundled ?checking for pjsip_dlg_create_uas_and_inc_lock in -lpjsip... no checking for pjsip_tsx_create_uac2 in -lpjsip... no checking if "pjmedia_mod_offer_flag flag = PJMEDIA_SDP_NEG_ALLOW_MEDIA_CHANGE" compiles using

Dear all, i have a problem with a system running - Ubuntu 10.04 ( all updates done ) - ii asterisk 1:1.8.5.0-1digium1~lucid Open Source Private Branch Exchange (PBX) - ii asterisk-dahdi 1:1.8.5.0-1digium1~lucid DAHDI devices support for the Asterisk PBX I also use freepbx 2.9 for the configuration. Hardware is a Dell R410 and a Digium

TE410P card down. I have three (3) TE410P in one machine running asterisk with SS7. My problems started last week when one of my cards started switching to E1 every time after reboot. I set the following in dahdi.conf and that solve the problem. /etc/modprobe.d/ options wct4xxp t1e1override=0x00 Now all 4 ports on that card is down with Red Alarm. I tried rebooting the machine and

Hi All, for a vicidial server which uses only voip, which is the minimum telephony card which would provide the required clock timing source for conferences to work properly ? Maybe the Digium TDM410PLF card without any daughter card would do the job ? Thank you very much for supporting. Have a nice week-end, Mike