search for: puzankin

...audio stream. 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 Urge...

...> 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

...T 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

...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

...h 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

...r: 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

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)

...ed 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 instead of...

...ed 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 instead of...

Hi, I need to get type of called number (TON), which is displayed in pri debug messages: Called Party Number (len=13) [ Ext: 1 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) 'xxxxxxxxxx' ] Does anyone know how to do it? According to documentation it is only possible for calling number. But I need to make decision in dialplan upon the value of type

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

...ed 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 line). (Repo...

...ed 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 line). (Repo...

Hi, I'm trying to connect Asterisk 1.4 and Cisco CallManager 5 using SIP Trunk without MTP (media termination point). Howerver, Cisco 79xx phones do not support RFC2833, they always notify CCM5 via SKINNY channel no matter where they send RTP to. For non-MTP trunk there's Out-of-band DTMF support in CCM5 called "kpml". I wonder if Asterisk can support it. I found an

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 -

...ed 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-24394 - CDR: FRACK with PJSIP directed pickup. (Reported by Richard Mudgett) * ASTERISK-24392 - res_fax: fax gateway sessions leak (Reported by Corey Farrell) * ASTERISK-24321 - SIP deadlock when running automated queues tests (Reported by Steve Pitts) * ASTERISK-2...

...ed 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-24394 - CDR: FRACK with PJSIP directed pickup. (Reported by Richard Mudgett) * ASTERISK-24392 - res_fax: fax gateway sessions leak (Reported by Corey Farrell) * ASTERISK-24321 - SIP deadlock when running automated queues tests (Reported by Steve Pitts) * ASTERISK-2...

The USB core was completely rewritten in 2.6, and as such the functions that ztdummy depends on do not exist in 2.6. I get the feeling that these changes are too much to easily fix ztdummy, so I don't expect to see it working on 2.6 any time soon (if ever) I made some small changes to zaprtc to work on 2.6 and I have MoH and Meetme functions working fine in my lab. For production I would

Hello Everyone, I've been playing with an agi script for paging sip phones. page.agi will take all available sip extensions and assign them to the global variable PAGE_GROUP. Allowing the phones to be paged from the dialplan with the new Page cmd. Extensions to be excluded are presented as arguments to the agi. Each time a page is made this agi refreshes the global variable. This works with

...y Corey Farrell) * ASTERISK-24237 - CDR: FRACK With PJSIP blonde transfer. (Reported by Richard Mudgett) * ASTERISK-24394 - CDR: FRACK with PJSIP directed pickup. (Reported by Richard Mudgett) * ASTERISK-18923 - res_fax_spandsp usage counter is wrong (Reported by Grigoriy Puzankin) * ASTERISK-22791 - asterisk sends Re-INVITE after receiving a BYE (Reported by not here) * ASTERISK-13797 - [patch] relax badshell tilde test (Reported by Tzafrir Cohen) * ASTERISK-24325 - res_calendar_ews: cannot be used with neon 0.30 (Reported by Tzafrir Cohen) * ASTERISK-...