[asterisk-users] Local channel scenario flushes CDR before dialplan end
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@office/n,,g) same = n,Noop(Notice the option /n and flag g, which allows to continue the dialplan after a destination channel hangs up, even it was transfered by a connected peer - it is very important for me) same = n,Noop(process some data, ask caller to value quality of service - another IVR, record some messages) same = n,Hangup() exten = h,1,Noop(I'm using func_odbc to save quiz results into DB, process recorded files, etc.) same = n,Noop(I'm using cdr_adaptive to store custom fields in table columns) same = n,CDR(my_custom_field_a)=my_value same = n,CDR(my_custom_field_z)=my_value [office] exten = 555,1,Dial(SIP/555) same = n,Hangup() A call comes from a SIP trunk directly to 555@incoming. It forks new pair of Local channels, bridging other leg to SIP/555. SIP peer answers the call, then hangs up. Dialplan continues right after Dial(Local/...). Also it goes to h extension after reaching Hangup in 555@incoming. Everything looks good, but CDR custom fields are empty, regardless that verbose shows that they were set in dialplan. After a short investigation I found out that CDR is written to DB in the same time when dialplan exits Dial application. It produces to records: SIP trunk to Local and Local to SIP/555, which is correct. 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. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk 1.8 SIP register uri: peer field ?
It's a peer name defined below in sip.conf. You may skip secret if it is specified in peer section. I don't know of any other meanings. For example, register = mypeer?u...@host [mypeer] type=peer defaultuser=user secret=blah ... This syntax exists since 1.6.2. 21.10.2010 17:31, Guillaume Bour пишет: Hello, Looking the asterisk 1.8 API documentation (http://www.asterisk.org/astdocs/api/index.html), I see a lot of new fields for sip register uris: register = [peer?][transport://]us...@domain][:secret[:authuse...@host[:port][/extension][~expiry] But the *peer* is not explained anywhere. What it is for ? Regards, Guillaume Bour. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.8 and Dial(SIP/peer_name) to undefined peer
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 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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MixMonitor with non-20ms packets
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 found nothing related to this issue. Is it a bug or result of misconfiguration? -- Best regards, Grigoriy Puzankin ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CDR and call transfer
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 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 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR and call transfer
I did a lot of googling until I found this thread: http://lists.digium.com/pipermail/asterisk-dev/2007-May/027666.html -- Grigoriy Puzankin Grigoriy Puzankin wrote: 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 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 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP kpml DTMF support in *
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 intertnet-draft for kpml: http://tools.ietf.org/id/draft-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. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP kpml DTMF support in *
Dan Austin wrote: If you are running the phone loads that shipped with CCM5, then your skinny phones have 'support' for RFC2833. CCM figures out during the call if the call will traverse a SIP trunk and instruct the phone to use RFC2833 for DTMF I have a CCM5-Asterisk trunk setup for MeetMe conferencing with NO MTP and DTMF works fine. Can you specify the version of the loads? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re[2]: [Asterisk-Users] Asterisk h323
On my Asterisk server it (chan_h323) gets 2-3 deadlocks every hour regardless of openh323/pwlib and asterisk versions (since the channel_h323 was not updated for a long time). The load is about 25-30 simultaneous calls (from h323 to zaptel, IAX and SIP). I have another Asterisk server. There's about 5-7 simultaneous calls, and deadlocks don't occur (calls go from zaptel to h323). AS Im using several Asterisk Box with chanh323 from asterisk, and it works AS fine. AS Sometime it gets deadlocks , but on 1.2.9.1 and 1.2.7 i have estability. AS A fail (crash) last month with about 600 calls per day. AS Regards AS Alberto Sagredo -- Grigoriy Puzankin ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DeadAGI and Hangup on channel
Hello,
GP I'm trying to catch channel hangup in DeadAgi script.
Googling didn't help. Channel status AGI command returns 6 - line is
up, because hangup has been requested only (and not completed).
I found the following solution. In res_agi.c source I found usage of
ast_check_hangup(chan) function. Then I added one AGI command to
retrieve ast_check_hangup function value.
Add the following routine to res_agi.c:
static int handle_getchannelhangup(struct ast_channel *chan, AGI *agi, int
argc, char *argv[])
{
int res;
res = ast_check_hangup(chan))
fdprintf(agi-fd, 200 result=%d\n, res);
return RESULT_SUCCESS;
}
Declare agi command usage:
static char usage_getchannelhangup[] =
Usage: GET CHANNEL HANGUP\n
\tChecks if hangup was requested on channel.\n
Returns 1 if hangup was requested on the channel, otherwise - 0.\n;
Add function definition to static agi_command commands[MAX_COMMANDS]:
{ { get, channel, hangup, NULL }, handle_getchannelhangup, Checks if
hangup requested on current channel., usage_getchannelhangup },
I'm not C programmer, my apologies to gurus.
If anyone find it useful then this could be a FEATURE REQUEST.
--
Grigoriy Puzankin
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[Asterisk-Users] DeadAGI and Hangup on channel
Hello, I'm trying to catch channel hangup in DeadAgi script. For example, A calls to DeadAgi script which connects (Dial) to B. After Dial command exits I need to identify where hangup came from: A or B. CHANNEL STATUS returns 6 (Line is Up) regardless of who hungup. In CLI show channels states that channel A to DeadAgi is UP even if A and B hungup. If A stays on the line after conversation with B (hangup from B), then DeadAgi would continue (with prompts and etc.), if A is off then DeadAgi should exit gracefully (not killed as Agi). Does anyone know how to do it? Thanks in advance. -- Grigoriy Puzankin ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
