all, I seem to be having a strange problem with a sip trunk. On a fairly frequent basis, Ill make a call, ore receive a call, and there will be NO sound.The strange part is that both endpoints are public IP addresses so NAT isnt in play and a snif..
pstn pstn asterisk link between avaya pbx both systems tied together..
What I am needing to do is to trim the 1 from beginning of the RDNIS and I have tried using the CUT function but cannot seem to make it work for me. What we have is a phone number like this, 18881232342 and want to make it like this 8881232342. I appreci..
has anyone experience with auto provisioning IP-phones on different locations through a central public provisioning server ? You use http or https ? Is there a danger that one uses a different MAC-address in the provisioning link to obtain SIP usern..
I am trying to configure a channel bank with 24 ports of FXS., but appear to be hitting a roadblock? This worked on v1.4.xx but now just get SimpleSwitch and immediate=no/yes dont seem to make a difference?, no matter if under top section, under chann..
, I installed 2 HB8 cards each of them with a Quad Bri modules in a HP 360 G6 running Debian Squeeze. Here is an output of dmesg wafter server has booted: [9.784123] wctdm24xxp 0000:0b:08.0: PCI INT A -> GSI 31 (level, low) -> IRQ 31 [ 11.847073] bn..
list, I have this problem with dropped calls on Asterisk. The setup is SIP internal extensions (Grandstream GXP-2000), two internal analogue DAHDI extensions and IAX2 trunk lines. IAX2 trunks use ulaw/alaw. The Internet connection is ADSL. Asterisk..
Im planning to use SGM with Asterisk, it is a commercial product. What is the different between SGM and libs77 and chan_ss7 ? Should I use SGM ? ________________________________ From: Tzafrir Cohen To: firstname.lastname@example.org Sent: Tue, Octo..