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...am looking to keep this operation within the SIP space, but if I have to go with SCCP to get this module working (if there is a way to make the phone treat the lines as protocol independent). Does anyone have any suggestions (or examples) as to how to accomplish this? Thanks. Eric -- Eric Lubow LinkExperts, Inc. Systems Administrator e: elubow@linkexperts.com w: www.linkexperts.com

...t always says, "You have an incoming call from number unavailable." And again, even if the callerID doesn't come up, the number is there 99% of the time. I have a feeling that all this interconnected somehow. Any help would be greatly appreciated. Thanks. Eric -- Eric Lubow LinkExperts, Inc. Systems Administrator e: elubow@linkexperts.com w: www.linkexperts.com

hi, I am facing some issues while using MixMonitor. My extensions logic is attached below: exten => s,1,MixMonitor(${CALLERID(number)}-${TIMESTAMP}-${UNIQUEID}.gsm,b) in this extensions TIMESTAMP is not working in Asterisk 1.4. can any help me why TIMESTAMP is not working in Asterisk 1.4. regards, Asif

I am getting this error [Aug 6 15:28:26] WARNING[24452]: chan_sip.c:1920 retrans_pkt: Maximum retries exceeded on transmission 8f68421-22821e1e at localhost for seqno 102 (Critical Response) [Aug 6 15:28:26] WARNING[24452]: chan_sip.c:1944 retrans_pkt: Hanging up call 8f68421-22821e1e at localhost - no reply to our critical packet. any Ideas? Jason

Hi, Has anyone here ever used a Polycom IP 4000 Soundstation SIP Conference Phone with asterisk? If so, how well does it work and how does it sound?

I am running Asterisk 1.4.5 and addons 1.4.1 in a CentOS 5 Server. My PBX has experienced several core dumps the last couple of days and I am not sure if this is what's causing it, but it always seems to happen when a particular extension on a grandstream phone uses ChanSpy SIP group. I have not been able to locate where the core dump file is being saved. I can't find it in my

All, I've a question on A*k hardware. I'm running 1.2.18 on a Dell DC051 (Intel(R) Celeron(R) CPU 2.80GHz) with 512mb RAM. I'm supporting 60 users (Cisco 7940s each + Xlite PCs). Call loads are low, max of about 10 simultaneous SIP/IAX calls. CPU for A*k rarely goes above 2% as I can tell. Its IP only, no E1/T1 cards present. However, I get complaints of bad voice quality,

All, I am having a lot of trouble with the Cisco 7961G phones. I have managed to get them up and running with Asterisk to the point where I can get incoming calls and make outgoing calls. The problem is when I make outgoing calls or extension to extension calls, the calls die after 20 seconds. I have google'd around and came up with little that is of help. The firmware version I am