similar to: H.323: Inbound calls, incorrect remoteIpAddress

Hello, I'm using digium E1 cards and serving SIP users at Asterisk. After the following call (see below) CDR shows two records. First looks as outbound call, but the second - as inbound call. Is it a bug or intended behavior? Call flow: SIP (ext: 100) -> ZAP (national number) SIP (ext: 100) transfers to SIP (ext: 200) SIP (ext: 200) -> ZAP (national number). In CDR it looks like

Hello, I'm trying to connect Cisco Call Manager 3.3 with Asterisk using H.323 Gateway. When I place call from a SIP phone registered at Asterisk to SCCP phone at CCM I can hear the voice in both directions. But when I call from SCCP phone at CCM to SIP phone at Asterisk the voice goes from CCM to Asterisk only. All devices have real IP-addresses - no NAT is used. Asterisk console does not

Hi, Is there something wrong with REALTIME (ARA) when used with rtcachefriends parameter? In my sip.conf (Asterisk 1.2.0): rtcachefriends=yes rtupdate=yes rtautoclear=yes Desired configuration is realtime configuration (via odbc) for SIP phones + MWI. Realtime means the following: when I make changes to db they should apply with no extra commands executed in CLI. In order to use MWI with

Hi, MixMonitor saves partial conversation when non-standard voice packet size is set (Asterisk 1.4.18.1). For example, if SIP-peer has alaw:30 then saved file would contain only 67% of total conversation. 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

Hi, In Asterisk 1.8.0 dialplan command Dial(SIP/peer_name) produces errors if no such peer_name defined instead of just saying "peer not found": [Nov 19 20:01:23] ERROR[7827]: netsock2.c:245 ast_sockaddr_resolve: getaddrinfo("sdf", "(null)", ...): Name or service not known [Nov 19 20:01:23] WARNING[7827]: chan_sip.c:5041 create_addr: No such host: sdf [Nov 19

The Asterisk Development Team has announced the release of Asterisk 1.8.32.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 1.8.32.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: Bugs

The Asterisk Development Team has announced the release of Asterisk 1.8.32.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 1.8.32.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: Bugs

The Asterisk Development Team has announced the release of Asterisk 11.14.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 11.14.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: Bugs

The Asterisk Development Team has announced the release of Asterisk 11.14.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 11.14.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: Bugs

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

The Asterisk Development Team has announced the release of Asterisk 12.7.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 12.7.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: Bugs

The Asterisk Development Team has announced the release of Asterisk 12.7.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 12.7.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: Bugs

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

Good day time! Using openssh with a serial attached terminal figures with a problem with flow control when serial terminal has printer connected to it etc. because ssh disables xon/xoff on associated terminal. At present ssh client always disables xon/xoff on associated terminal device, regarding of it's previous state, e.g. ixon and ixoff options were set or not. By searching the google

Hi, There's a quite complex dialplan scenario and I found out that CDR of main channel is flushed right after hangup on Local channel. I will try to simplify my scenario: [incoming] exten => 555,1,Noop(do something before using local channel, fill some variables, play IVR menus and so on) same => n,Dial(Local/555 at office/n,,g) same => n,Noop(Notice the option "/n" and

Hi, Is it possible to assign hint extension to Local channel? 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)

Hi all Follow up to this post, we've been able to capture a gdb backtrace. Can anyone help with guidance as to what this means. See below: (gdb) bt #0 0xffffe410 in ?? () #1 0x00000001 in ?? () #2 0x00000000 in ?? () #3 0xbfffc9d8 in ?? () #4 0x402b36e3 in __waitpid_nocancel () from /lib/tls/libc.so.6 #5 0x4025ef58 in do_system () from /lib/tls/libc.so.6 #6 0x402268dd in system ()

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

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 -