search for: grigoriy

...ream. Is it a bug in h.323? Is there some more settings to make in .conf files? See detailed debug below: *CLI> == New H.323 Connection created. -- Received SETUP message -- Setting up Call -- Call token: [ip$195.128.54.2:2689/1] -- Calling party name: [Puzankin Grigoriy] -- Calling party number: [5522] -- Called party name: [822] -- Called party number: [822] Urgent handler =-= In OnAnswerCall for call 1 Urgent handler We're at 195.128.54.20 port 15500 Urgent handler Answering/Requesting with root capability 4 Urgent handle...

...t: 100) -> ZAP (national number) SIP (ext: 100) transfers to SIP (ext: 200) SIP (ext: 200) -> ZAP (national number). In CDR it looks like SIP (ext: 100) -> ZAP (national number) ZAP (national number) -> SIP (ext: 200) How to identify the second CDR as outbound call? Best regards, -- Grigoriy Puzankin

...According to documentation it is only possible for calling number. But I need to make decision in dialplan upon the value of type of called number. BTW, I made a little research on source code and could not find anything related to my question. Perhaps, it's not implemented. Best regards, Grigoriy -- ? ?????????, ???????? ????????

...raft-ietf-sipping-kpml-07.txt, but it seems to be very old - "Expires June 25, 2005". I know that using MTP in SIP Trunk at CCM5 makes DTMF work in RFC2833, but MTP resource is very limited and I don't want to proxy RTP via CCM5. Please, do not offer to use H.323. Thanks in advance. Grigoriy.

I know it's a common error and there is a lot of help available but still can't resolve the issue: all i am trying to do is to read a csv file from my folder and this is what i get: Error in scan(file, what, nmax, sep, dec, quote, skip, nlines, na.strings, : scan() expected 'a real', got '1,m,a,F,165,240,26.5,31,0.738,0.704,1.095,0.606,0.847' Can you help? -- View

Hello, I got some of the R source code and not being able to Run it in RStudio. I get the error: Error in setwd("dir") : cannot change working directory I have gone through forums but nothing seemed relevant to my issue. What bugs me the most is the ("dir") that the error shows, is of those who wrote the source code and not mine(it still sees the directory of their

...quests to enable "xon-xoff" after the client's request of pty allocation succeeds. Now we can use printers attached to serial terminals and other devices. In spite of the reason, that patch works well for me, i suppose it might be better but i need developers advice. Best wishes, Grigoriy A. Sitkarev -------------- next part -------------- A non-text attachment was scrubbed... Name: openssh-4.5p1_localflow.patch Type: text/x-patch Size: 6127 bytes Desc: not available Url : http://lists.mindrot.org/pipermail/openssh-unix-dev/attachments/20061209/24d3faab/attachment.bin

...s - no NAT is used. Asterisk console does not report any errors with H.323. Does anyone know how to cope with this problem? Summary: SIP phone <-> Asterisk <-> H.323 <-> CCM <-> SCCP phone SCCP phone -> CCM -> H.323 -> Asterisk -> SIP phone -- Grigoriy Puzankin

...with old password - OK. This is not REALTIME behavior! From my opinion caching should not be used for all kind of events - it should be used for events like MWI notification, finding IP address to route incoming calls to, and so on, but NOT for making outgoing calls and for register requests. -- Grigoriy Puzankin

...tion. With alaw:20 MixMonitor saves 100% of conversation. It seems that MixMonitor has hardcoded "packets per second" or "samples per packet" values. I did a lot of googling, but found nothing related to this issue. Is it a bug or result of misconfiguration? -- Best regards, Grigoriy Puzankin

...reate_addr: No such host: sdf [Nov 19 20:01:23] NOTICE[7827]: channel.c:5106 __ast_request_and_dial: Unable to request channel SIP/sdf I didn't find any bug report regarding this issue. Is there any setting in sip.conf to disable host resolving in case of undefined peer name? -- Best regards, Grigoriy Puzankin

...ect. If I use SIP channel instead of Local, then CDR is written after dialplan ends and all fields are set. But in this case I loose call processing after it was transfered to another party (I have a lot of contexts - catching a call-end is a pain). Is it a bug or intended behavior? Best regard, Grigoriy.

...annel? Something like this: exten => 555,hint,Local/123123123 at my-context The purpose is to subscribe to this channel state from SIP-phone. I know that queues can track Local channel status, however I could not find any information regarding using Local channel in hints. -- Best regards, Grigoriy

Hi, I'm having a problem calling international numbers with debian's Asterisk 1.0.7 w/ zaptel 1.0.9 in Moscow. Russia doesn't seem to have touchtone dialing, so pulsedial is enabled on my TDM400P interface. Local numbers work fine, but when it comes to long distance or international, I'm lost. The prefix for these should be 8 (wait for dialtone) 10 (country code) (city code)

...(Reported by Tzafrir Cohen) * ASTERISK-13797 - [patch] relax badshell tilde test (Reported by Tzafrir Cohen) * ASTERISK-22791 - asterisk sends Re-INVITE after receiving a BYE (Reported by Paolo Compagnini) * ASTERISK-18923 - res_fax_spandsp usage counter is wrong (Reported by Grigoriy Puzankin) * ASTERISK-24393 - rtptimeout=0 doesn't disable rtptimeout (Reported by Dmitry Melekhov) * ASTERISK-24063 - [patch]Asterisk does not respect outbound proxy when sending qualify requests (Reported by Damian Ivereigh) * ASTERISK-24425 - [patch] jabber/xmpp to use TLS inst...

...(Reported by Tzafrir Cohen) * ASTERISK-13797 - [patch] relax badshell tilde test (Reported by Tzafrir Cohen) * ASTERISK-22791 - asterisk sends Re-INVITE after receiving a BYE (Reported by Paolo Compagnini) * ASTERISK-18923 - res_fax_spandsp usage counter is wrong (Reported by Grigoriy Puzankin) * ASTERISK-24393 - rtptimeout=0 doesn't disable rtptimeout (Reported by Dmitry Melekhov) * ASTERISK-24063 - [patch]Asterisk does not respect outbound proxy when sending qualify requests (Reported by Damian Ivereigh) * ASTERISK-24425 - [patch] jabber/xmpp to use TLS inst...

I thought this had been around before but I can't seem to find anything about it. I have a customer whom prior to upgrading to Asterisk invested in one of those boxes that plays your company sales campaign into the MOH port on your key system. For reasons of message maintenance he wants to keep the box as part of the new system. Can I couple this to the sound card in the Asterisk server

...(Reported by Tzafrir Cohen) * ASTERISK-13797 - [patch] relax badshell tilde test (Reported by Tzafrir Cohen) * ASTERISK-22791 - asterisk sends Re-INVITE after receiving a BYE (Reported by Paolo Compagnini) * ASTERISK-18923 - res_fax_spandsp usage counter is wrong (Reported by Grigoriy Puzankin) * ASTERISK-24392 - res_fax: fax gateway sessions leak (Reported by Corey Farrell) * ASTERISK-24393 - rtptimeout=0 doesn't disable rtptimeout (Reported by Dmitry Melekhov) * ASTERISK-23846 - Unistim multilines. Loss of voice after second call drops (on a second lin...

...(Reported by Tzafrir Cohen) * ASTERISK-13797 - [patch] relax badshell tilde test (Reported by Tzafrir Cohen) * ASTERISK-22791 - asterisk sends Re-INVITE after receiving a BYE (Reported by Paolo Compagnini) * ASTERISK-18923 - res_fax_spandsp usage counter is wrong (Reported by Grigoriy Puzankin) * ASTERISK-24392 - res_fax: fax gateway sessions leak (Reported by Corey Farrell) * ASTERISK-24393 - rtptimeout=0 doesn't disable rtptimeout (Reported by Dmitry Melekhov) * ASTERISK-23846 - Unistim multilines. Loss of voice after second call drops (on a second lin...

Hello everyone. I have a timing slips errors and I can't understand what source of the problem is. My installation has 2 digium cards: TE420 and TE220 cards in one server. There are 3 spans (E1) to PSTN and 3 spans to internal PBS stations - normal installation for transit communication. Span configuration is: span=1,1,0,ccs,hdb3 #TE420 - first port. To PSTN. span=2,0,0,ccs,hdb3 #TE420 -