[Asterisk-Users] fax with asterisk
GFI MailSecurity's HTML threat engine found HTML scripts in this email and has disabled them. From couple of weeks i am working on asterisk fax but was not successful. I am able to receive only half fax documents and its ending with blurred lines. I have tried with almost all the versions of spandsp 0.0.2pre4,pre10 etc but of no use. I have digium card to which i have plugged in 2 pstn lines and i have removed echocancellation too(zapata.conf) and also enabled g7111 alawUlaw codecs. When ever i receive a fax it will come upto some extension(half) and ends.I even checked the pstn lines but its working fine. Please look into it and help me to get rid off. BELOW ARE THE LOGS I HAVE GOT : = Spawn extension (zapincoming, fax, 0) exited non-zero on 'Zap/2-1' Changed from phase 0 to 1 Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down 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 20 20 20 20 20 20 20 20 DIS: Preferred octets: 256 Can receive fax Supported data signalling rates: V.27ter and V.29 R8x7.7lines/mm and/or 200x200pels/25.4mm OK 2D coding OK Scan line length: 215mm Recording length: A4 (297mm) Receiver's minimum scan line time: 0ms at 3.85 l/mm: T7.7 = T3.85 R8x15.4lines/mm OK Minimum scan line time for higher resolutions: T15.4 = T7.7 DIS: 80 00 ce f0 80 80 01 HDLC underflow in state 9 Changed from phase 4 to 3 Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up TSI: 43 35 32 30 39 35 33 33 32 20 20 20 20 20 20 20 20 20 20 20 20 TSI without final frame tag Remote fax gave TSI as: XX DCS: 83 00 c6 f0 80 80 00 DCS with final frame tag In state 9 DCS: Can receive fax Selected data signalling rate: V.29, 9600bps R8x7.7lines/mm and/or 200x200pels/25.4mm OK 2D coding OK Scan line length: 215mm Recording length: A4 (297mm) Minimum scan line time: 0ms Minimum scan line time for higher resolutions: T15.4 = T7.7 Get at 9600 Changed from phase 3 to 5 Fast carrier up Training failed (sequence failed) Fast carrier training failed Fast carrier down Fast carrier up Coarse carrier frequency 1699.76 (66) Training error 15.399144 Training succeeded (constellation mismatch 10.791423) Fast carrier trained Fast carrier down Changed from phase 5 to 4 Start rx document - compression 2 Start rx page CFR: 84 HDLC underflow in state 5 Post trainability Changed from phase 4 to 5 Fast carrier up Fast carrier down Fast carrier up Coarse carrier frequency 1699.81 (66) Training error 3.356434 Training succeeded (constellation mismatch 2.404210) Fast carrier trained Feb 11 19:27:21 NOTICE[20570]: chan_sip.c:7531 handle_request: Fast carrier down Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 955 (got 1906, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 962 (got 2596, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 963 (got 1731, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 965 (got 2176, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 967 (got 1730, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 969 (got 2657, expected 1728). Fax3Decode2D: Warning, (FakeInput): Premature EOL at scanline 971 (got 1723, expected 1728). Fax3Decode2D: Warning, (FakeInput): Premature EOL at scanline 972 (got 1723, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 973 (got 1729, expected 1728). Fax3Decode2D: (FakeInput): Uncompressed data (not supported) at scanline 974 (x 1340). Fax3Decode2D: Warning, (FakeInput): Premature EOL at scanline 974 (got 1340, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 975 (got 1741, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 976 (got 1729, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 979 (got 1795, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 980 (got 1795, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 981 (got 2032, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 982 (got 2699, expected 1728). Fax3Decode2D: (FakeInput): Uncompressed data (not supported) at scanline 983 (x 262). Fax3Decode2D: Warning, (FakeInput): Premature EOL at scanline 983 (got 262, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 984 (got 2026, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 986 (got 1731, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 987 (got 1730, expected 1728). Fax3Decode2D: Warning,
[Asterisk-Users] Dialogic Support
I am a newbie to asterisk pbx. I got a dialogic card with 2 ports. Can any one tell whether asterisk supports dialogic cards? Update me if you have info about the drivers installation and support. Thanks Regards V.Venu ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Unable to get Incoming Calls
Till yesterday my asterisk worked fine. Today i am not getting any incoming calls to my asterisk box. I dont have any problem in calling outside but only with incoming even with the same configuration. Can any one help me to get rid off? winmail.dat___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Music On Hold Problem
Recently I have configured Music On Hold option in asterisk PBX. But I am unable to listen to the audio properly and morever its getting breaks for every 3 seconds. If any one know about this. Please help me Thanks Regards V.Venu -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Sunday, November 14, 2004 12:07 PM To: [EMAIL PROTECTED] Subject: Asterisk-Users Digest, Vol 4, Issue 181 Send Asterisk-Users mailing list submissions to [EMAIL PROTECTED] To subscribe or unsubscribe via the World Wide Web, visit http://lists.digium.com/mailman/listinfo/asterisk-users or, via email, send a message with subject or body 'help' to [EMAIL PROTECTED] You can reach the person managing the list at [EMAIL PROTECTED] When replying, please edit your Subject line so it is more specific than Re: Contents of Asterisk-Users digest... Today's Topics: 1. RE: SysMaster and GPL Violation (Brian) 2. Re: getting callerid from spa3k to asterisk (Randy Bush) 3. my asterisk drops connection when remote side putsme on hold? (Steve Prior) 4. Cisco ATA and G729 (kido noagbodji) 5. Remote answer not detected (DB) 6. Re: SysMaster and GPL Violation (Voip Business) 7. RE: Cable for T1 connection: Crossover or straightthrough? (Franceen Thompson) 8. RE: Cisco ATA and G729 (Franceen Thompson) 9. Queue/AgentCallbackLogin Problems (Franceen Thompson) -- Message: 1 Date: Sat, 13 Nov 2004 19:30:06 -0700 From: Brian [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] SysMaster and GPL Violation To: 'Asterisk Users Mailing List - Non-Commercial Discussion' [EMAIL PROTECTED] Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii Are you saying that those of us that are using the product should not be allowed to voice our opinions about its licensing, development and maintenance? That we should all just shut up and take whatever Mark co. give us? If that's the case, then this is most definitely NOT an open-source project at all. -Original Message- From Brandon Patterson Sent: Saturday, November 13, 2004 7:15 PM Uh ok...So when will Asterisk be a licensed product? Will it take the form of a Redhat sort of platform... Fedora with Redhat the pay me money side of the house? Just a simple question: When can we expect to see Asterisk the licensed as in paid for version ? Brandon Right now. As far as I know, you just need to contact Digium's sales department and negotiate a licensing agreement with them. -- Message: 2 Date: Sat, 13 Nov 2004 19:11:10 -0800 From: Randy Bush [EMAIL PROTECTED] Subject: [Asterisk-Users] Re: getting callerid from spa3k to asterisk To: splatters [EMAIL PROTECTED] Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii if i have two sip contexts for my spa3k, on inbound and one outbound, e.g. [spa3k-in] type=friend host=dynamic port=5061 auth=md5 secret=pfui qualify=1000 canreinvite=yes context=ext-in42 [spa3k-out] type=peer auth=md5 secret=pfui username=outpass fromuser=outpass host=spa3k.bogus.com port=5061 nat=no canreinvite=yes context=ext-in42 and the spa3k's PSTN / Subscriber Information / User ID: = spack-in, the incoming connection from spa3k to * is being routed to the spa3k-out context, not the spa3-in context. see appended. i suspect this is a bug in * 1.0.1. i found the problem, or at least a work-around. if i reverse the order of the above two sip contexts, the incoming call is properly routed to the spa3k-in sip context as opposed to the wrong one, spa3k-out. my guess is that * is traversing a list and taking the first context which has the ip address and port it wants without checking the context name against the name which was received over the wire. so it depends on what order the contexts are inserted in the list. aii! randy -- Message: 3 Date: Sat, 13 Nov 2004 22:33:58 -0500 From: Steve Prior [EMAIL PROTECTED] Subject: [Asterisk-Users] my asterisk drops connection when remote side puts me on hold? To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii; format=flowed I've got a TDM100P card with a fxo and fxs module in the US. I'm using kewlstart for all ports. I've noticed that when I make a call out from an analog phone out the POTS line that if after talking to the party I called (in this case the phone company itself) they put me on hold asterisk disconnects the call immediatly. I've looked around the web pages, but can't figure out what might be causing this and how to fix it - can anyone give me a clue? Thanks
