similar to: Using * and other gateways together

Dear Folks, Anyone knows if Sangoma supports or going to provide support for battery removal detection on FXO lines?? As Tzafrir said earlier DAHDI supports it, which is a very nice feature but what about Sangoma? Regards. -- M. Shokuie Nia. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Dear folks, I have problem in fax reception. The astrisk detects the fax tone and jusmps to the fax extension and rxfax application starts and the max machine starts the fax but saddenly stops and seems the rxfax have died. It doesnt returns, not files in the output dir and .......... Anyone have any idea or help how could i get whats wrong here. Regards. M. Shokuie Nia

Dear folks, I have a problem with console/dsp using ALSA. I dont know why the output sound is choppy sometimes and also the input one has an awful delay. Is there anyone here with experince about ALSA channels or not? I would be highly appreciated if anyone could help me. Regards. M. Shokuie Nia _________________________________________________________________ Don't just search. Find.

Dear folks, I got a problem sending faxes using spandsp. Primerily, when the tiff file made using GIMP 2 with different compresions the fax app break downs whole *. Moreover when i made a tiff file using Microsoft mdi, everything works fine but on the other end of the call, the received fax is shrinked in size. Anyone has any idea whats the right file format and compression type for it? PS.

Dear folks, I couldnt receive faxes and get the following debug traces on the console, I appreciate any help or even hints. Using Spandsp-0.2 app_rxfax.c:76 span_message: FLOW Get at 9600bps, modem 1 app_rxfax.c:76 span_message: FLOW Changed from phase 3 to 5 app_rxfax.c:76 span_message: FLOW Non-ECM carrier up app_rxfax.c:76 span_message: FLOW Non-ECM carrier down app_rxfax.c:76 span_message:

Hi folks, I had a survey online but there i couldnt find a clean sample of CAS signalling on E1 interfaces. I defined a span with CAS framing and HDB3 line coding but dont know which signalling to use for channels. I'd use 3 bit CAS signalling and 20 incoming channels and 10 outgoing ones. Anyone could help me define the signalling for these channels. PS. Im using Sangoma cards. Any

Dear folks, I would be very thankful if an experienced user can help me out here. I wanna use mfcr2 and unicall library on sangoma boards but so far impossible for me. As Im setting the framing type to CAS (TE_SIGMODE = CAS) on wanpipe I couldnt get the link alarm out (i looped a A102d links) but when setting it to CCS everything works fine and the green lights shine on the back. Can anyone

Dear folks, I'm using * realtime with no problems on most of the systems i've setup but rarely i confront this problem that the asterisk doesn't load from database when the systems rebooted and i have to reload it manually or restart it, but it would work fine afterward, no problem how many times you stop and start the *. It seems, there is a missequence of deamon loading at boot

Hi folks, Its the forth day I'm sticking to a problem with chan_alsa, The sound played or captured from the device is choppy time to time. I mean when talking on a console/dsp microphone the other side hear my sound choppy and I'm hearing hers the same but not all the time during a call, sound sometimes are clear. Even when I'm putting the sip side on hold i hear the same choppy music

Hi everyone, We are a longtime samba/win desktop shop but are getting more Macs. Samba sernet rpms 3.3.9 on CentOS 5.4 test box (started to test 3.43 but will wait) and I confirmed a problem I have been having in production. Server is ADS member with ldap stored idmap. All AD auth and idmap are identical to production servers and have been working fine for years. Filesystem is XFS and there are

Hello All, I am trying to setup a small system where Nextone Softswitch will send traffic to Asterisk and then terminate on Grandstream GXW410x IP Analog Gateway but for some odd reasons the call are flashed back from Grandstream to Asterisk and creating a Black loop... I did follow the instructions provided by Grandstream support but it doesn't seems to be working...

Hi! I have gone a bit further with the installation of Python with Wine. I have downloaded http://www.dll-files.com/dllindex/dll-files.shtml?msvcirt and placed it in the .wine/drive_c/windows/system directory. And I have run msiexec /i python-2.4.msi within the .wine/drive_c directory. The Python setup windows pops up but when selecting the "Destination directory" I get the

Hello, My company is doing some * testing with our Class 5 softswitch and had some questions regarding ringback being provided to our PSTN users (off --> on net calling) Currently with MGCP subscribers, we know the PSTN ringing is provided by a digital PBX for example, However, it looks like with SIP, our softswitch is relying on MGCP signaling on our PSTN gateways to provide ringback

help me in another issue related also to registering asterisk with another softswitch: A) If nat=yes, then I have to set canreinvite=no to be able to register, correct? B) In case of using firefly softphone, how it possible to set it to have nat=yes (at the firefly it self and not at the sip user configuration section)? As most of the sip endpoint give an option to set nat=yes and so on, how it

Dear List; I am trying to create a link between Asterisk and My softswitch, the link to be SIP Trunk. I did the below configuration and I do not know if any one can help me and advise me to have better configuration to be sure that link is fine. But I do not know how to determine the SIP username to be sent for my softswitch as sometimes the softswitch need to check it. Also, does asterisk

I've been working in the VoIP industry for just a bit over a year now... Mostly taking care of the underlying systems. I've now reached the point where I'm being drawn more and more into the call processing side of things. My background is in computer and "classic" telephony systems (DMS250/MTX, DSC 400, T1, channel banks. telabs analog echo supressor modules and

Hello everybody, Im new to the list and also new to asterisk, Im wondering if I could set up asterisk as a softswitch, I guess for what I've been reading that It could be possible but almost all the info and documentation Ive found so far is about asterisk as a PBX, etc. Im willing to set a small voip wholesale traffic bussiness and Im not quite sure asterisk is the right chocie for that.

Hi All, I want to use Asterisk for VoiceMail for a softswitch. I can dial in to leave voicemail and retrieve. Now there are many SIP Endpoints registered to the Softswitch. The Asterisk is sending a NOTIFY msg to the Softswitch on <ip addr>:0 Somehow Asterisk Looses the port from where the INVITE came in, this NOTIFY msg is not going out of the Asterisk, I cannot see in Ethereal.

On 07/21/2015 02:30 PM, Binet, Valere (NIH/NIA/IRP) [C] wrote: > How about > > $ sudo yum list installed | grep <reponame> YES! This did it! Thank you! > > It won?t work in CentOS 5 but it should work in 6 or 7 > > Val?re Binet [C] > IT Security Administrator > Kelly Government Solutions On-Site at the NIH > NIH / NIA / IRP > Tel : 410 558 8013 >