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...w stack -- Goto (fax,699,1) -- Executing Macro("Zap/2-1", "faxreceive") in new stack -- Executing SetVar("Zap/2-1", "FAXFILE=/var/spool/asterisk- fax/1087447538.0.tif") in new stack -- Executing SetVar("Zap/2-1", "EMAILADDR=ipso@snappymail.ca") in new stack -- Executing RxFAX("Zap/2-1", "/var/spool/asterisk-fax/1087447538.0. tif") in new stack Changed from phase 0 to 1 Slow carrier up Start receiving document Changed from phase 1 to 4 Sending ident >>> CSI: 40 20 20 20 20 20 20 20 20 20 20 20 20...

Hello, We just had some new T1s turned up today to replace others that our contract has run out on and we are now getting more echo on the new T1 lines than we had on the old ones. The T1 type is 24-channel, D4/AMI SF Robbed-bit(the same as the T1s they replaced) The problem is that we are getting echo on about 10% of the calls in and out placed on these new T1s compared to less than 1% with

Has anyone seen this problem before? I have a server with a single X100P card. The audio level is a low, but if I raise the gain to more than -2db (Rx + Tx) it starts to oscillate in an echo test. Not at a high frequency but with a noise that is best described as a steam engine starting up. It then starts to clip and crackle. If I bring the gain down to Rx=-2.0 and Tx=0.0 or lower then it settles

...1,Goto(reverse-hold,s,1) Ideally something like this would become its own application, or tied in to something like parkedcalls. But for now, the above works relatively well. The biggest drawback is there is no way to get back to call on hold until it times out. Enjoy. -- Mike Benoit <ipso@snappymail.ca>

...ans with the equipment necessary to help find the problem if the problem really is their outside plant. However, getting to that person can be a real challenge." Any suggestions on ways to overcome the challenge of getting the right technician on the phone? Thanks. -- Mike Benoit <ipso@snappymail.ca>

Hi ! I need a solution to park incoming calls to an extension of my choice where a special announcement is played, park subsequent calls to specific pools so that they listen to announcements of my choice. any ideas ? Shah.

...me issues. I wonder if when calls are made from SPA-2000's to PSTN numbers through Asterisk, asterisk is just amplifying the SPA-2000's own echo somehow. Any test results are welcome, I would be very interested if other people are unable to replicate my results. -- Mike Benoit <ipso@snappymail.ca>

...everything under the sun to get rid of echo on my X100P, I'm curious if anyone managed to solve the echo issues by switching to a SPA-3000? As well, if you have multiple SPA-3000's, can you create dial-out groups similar to the Dial(ZAP/g1) functionality? Thanks. -- Mike Benoit <ipso@snappymail.ca>

...y extension + #, it will ring a SIP phone no problem. So now I just need to get it to that automatically. The SPA-3000 User guide shows how to have it automatically forward incoming PSTN calls to its FXS port, but that would normally be a phone, not Asterisk. Any ideas? -- Mike Benoit <ipso@snappymail.ca>

I just added a second X100P card to my * server, altough it seems to be working * seems to be ignoring it: zaptel.conf: ----------------- fxsks=1-2 loadzone=us defaultzone=us zapata.conf: ------------------ context=inbound-analog signalling=fxs_ks group=1 channel => 1 channel => 2 I created a couple of test extensions: ; test extensions

...ype: application/sdp^M Content-Length: 214^M ^M v=0^M o=root 22051 22051 IN IP4 192.168.1.3^M s=session^M c=IN IP4 192.168.1.3^M t=0 0^M m=audio 14996 RTP/AVP 0 101^M a=rtpmap:0 PCMU/8000^M a=rtpmap:101 telephone- event/8000^M a=fmtp:101 0-16^M a=silenceSupp:off - - - -^M -- Mike Benoit <ipso@snappymail.ca>

...ax error configure:5144: error: C++ preprocessor "/lib/cpp" fails sanity check See `config.log' for more details. I'm running Mandrake 10.0, with the follow gcc packages installed: gcc-3.3.2-9mdk gcc-cpp-3.3.2-9mdk Any ideas how to fix this? Thanks. -- Mike Benoit <ipso@snappymail.ca> -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 189 bytes Desc: This is a digitally signed message part Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20041221/0cb4a1c4/attachment.pgp

...tp.c:429 ast_rtp_read: RTP: Received packet with bad UDP checksum Aug 19 15:53:44 NOTICE[1173711792]: rtp.c:429 ast_rtp_read: RTP: Received packet with bad UDP checksum Aug 19 15:54:04 NOTICE[1173711792]: rtp.c:429 ast_rtp_read: RTP: Received packet with bad UDP checksum -- Mike Benoit <ipso@snappymail.ca>

I am having trouble getting transfers to work when a zap channel is part of the call. I have a couple SPA-2000's and some X100P cards as my setup. This is what I'm trying: Dial number from phone: -- Executing Dial("SIP/206-2c61", "Zap/1/#######") in new stack Currently on call: -- Called 1/####### Press flash to place call on hold with SPA-2000: --

...39;:1027] Tx-Frame Retry[-01] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 00003ms SCall: 00003 DCall: 00001 ['Server 2':1027] Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: POKE -- Mike Benoit <ipso@snappymail.ca> -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 189 bytes Desc: This is a digitally signed message part Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20041221/d54e1d42/attachment.pgp

...e long (250ms) winks usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes musiconhold=default faxdetect=both echocancel=yes echocancelwhenbridged=yes echotraining=800 -- Mike Benoit <ipso@snappymail.ca>

I am considering using Sipura-3000s as FXO devices for my * system. Has anyone tried them in that configuration? They interest me because they need no PCI slots and therefore no drivers. I would much prefer not to have any special kernel requirements for my system. /carmi

Anyone have experience with Sipura's? Anyone know if they offer a warranty? Would like opinions on these, good or flame. We bought *one* to test with and it died, can't even get a response from Sipura "support". Could anyone recommend another device to replace these? Prefer 1 or 2 port design. Ty :-)

I created an account at IAXTEL.com to route 1-700-XXX-XXXX calls through. IAXTEL.com gave me a number (example) of 700-555-6226. I have made the following changes to my: /etc/asterisk/extensions.conf: [iaxtel700] exten => _81700XXXXXXX,1,Dial(IAX2/myusername:mypassword@iaxtel.com/${EXTEN:1}) exten => _81800NXXXXXX,1,Dial(IAX2/myusername:mypassword@iaxtel.com/${EXTEN:1})

...he results were pretty bad. --- Results after 384 passes --- Best: 36.779785 -- Worst: 1.562500 Is this a poor mainboard issue, or is it actually not possible to do IDE software RAID on a machine running Asterisk with X100P cards? Is anyone currently doing it? Thanks. -- Mike Benoit <ipso@snappymail.ca>