[asterisk-users] VoIP Information in CDRs
Hi,
I am trying to find a way to add the following info in CDRs (with
asterisk 1.4.23.1):
1. Codec used
2. RTP QoS statistics
3. RTP IP of remote host
4. For answered calls, the peer that requested to end the conversation
I have managed to get 1 and 2 for the caller, like that:
exten => h,1,Set(CDR(userfield)=RIP=${SIPCHANINFO(recvip)}
Codec=${CHANNEL(audioreadformat)}/${CHANNEL(audiowriteformat)}/${CHANNEL(audionativeformat)}/${SIPCHANINFO(t38passthrough)}
QOS=${RTPAUDIOQOS})
The problems I have so far:
*1. CODEC
*Codec is reported only for A-Leg.
When transcoding asterisk logs the above line as: slin for read / slin
for write / the codec of A-Leg / 0 for t.38.
Is there a way to get the codec for both legs of a call?
*2. RTP Qos is reported only for A-Leg.
*Also, asterisk seems to ignore the RTP statistics reports by B-Leg
after the BYE:
-- Executing [...@core-dialplan:3] Hangup("SIP/401-08231540", "") in
new stack
== Spawn h extension (core-dialplan, h, 3) exited non-zero on
'SIP/401-08231540'
Scheduling destruction of SIP dialog
'0aa4f73f5c9715b7661b50080a669...@10.11.12.1' in 6656 ms (Method: INVITE)
set_destination: Parsing for
address/port to send to
set_destination: set destination to 10.11.12.43, port 5060
Reliably Transmitting (no NAT) to 10.11.12.43:5060:
*BYE sip:4...@10.11.12.43:5060;transport=udp SIP/2.0
*Via: SIP/2.0/UDP 10.11.12.1:5060;branch=z9hG4bK3077b703;rport
From: "Office Line 1" ;tag=as1d9352fe
To:
;tag=0009b7aa1aaa51eb2c767e13-7fb3b34a
Call-ID: 0aa4f73f5c9715b7661b50080a669...@10.11.12.1
CSeq: 103 BYE
User-Agent: home.tsaousis.gr
Max-Forwards: 70
Content-Length: 0
---
== Spawn extension (core-dialplan, 422, 1) exited non-zero on
'SIP/401-08231540'
box*CLI>
<--- SIP read from 10.11.12.43:50539 --->
*SIP/2.0 200 OK
*Via: SIP/2.0/UDP 10.11.12.1:5060;branch=z9hG4bK3077b703;rport
From: "Office Line 1" ;tag=as1d9352fe
To:
;tag=0009b7aa1aaa51eb2c767e13-7fb3b34a
Call-ID: 0aa4f73f5c9715b7661b50080a669...@10.11.12.1
Date: Sat, 21 Feb 2009 14:29:42 GMT
CSeq: 103 BYE
Server: Cisco-CP7960G/8.0
Content-Length: 0
_*RTP-RxStat: Dur=4,Pkt=180,Oct=28800,LatePkt=0,LostPkt=0,AvgJit=0
RTP-TxStat: Dur=4,Pkt=183,Oct=29280
*_
These SIP messages are being exchanged after the dialplan has executed
the h extension.
Is there a way to have RTP statistics for both legs?
*3. RTP IP is not reported anywhere.
*The RIP= variable I have above, reports the SIP IP, and again only for
A-Leg.
Is it possible to find out the RTP (not SIP) IPs for both legs?
*4. Find out who hangedup an answered call.
*I have not found any way to determine the peer that requested to hangup
the call.
Is it possible to find who of the two legs requested the hangup?
Any help is appreciated.
Costa
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Re: [asterisk-users] TDM400P periodic sound clicks on FXS
Tzafrir Cohen wrote: > How about calls from either the card or the trunk to an echo test > extension? to a local SIP/IAX phone? > > It seems that the FXS slots do no have the issue with local VoIP phones. Any ideas? ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] TDM400P periodic sound clicks on FXS
Matthew Fredrickson wrote: Costa Tsaousis wrote: Hi, I am having periodic sound clicks (2-3 per second) on all FXS of a TDM400P when the remote end is my VoIP provider. However: - recording the conversation on the asterisk, does not have the glitches, although I can hear them on a real phone. - My VoIP provider to my VoIP phones through the same asterisk is OK. - TDM to TDM through the same asterisk is OK. If TDM to TDM is ok, then it would strongly point towards a problem with perhaps the VoIP provider. This is just shooting off of my hip, but maybe a jitterbuffer issue, like with the phones? I think when Asterisk bridges SIP->SIP calls, it doesn't do any jitter buffering. If it is a jitterbuffer, then why the recordings (of the same calls I hear the clicks, not other calls) do not have them? Also, I believe it cannot be an issue of the provider since all ATAs I tested do not have the issue (same provider, same account). ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] TDM400P periodic sound clicks on FXS
Hi, I am having periodic sound clicks (2-3 per second) on all FXS of a TDM400P when the remote end is my VoIP provider. However: - recording the conversation on the asterisk, does not have the glitches, although I can hear them on a real phone. - My VoIP provider to my VoIP phones through the same asterisk is OK. - TDM to TDM through the same asterisk is OK. I tried with and without echocancel and different values of echotrain (including 'no'), without luck. The card is not sharing interrupts. Any ideas? Kernel is 2.6.9 asterisk is 1.2.19-BRIstuffed-0.3.0-PRE-1y-h # lspci 00:00.0 Host bridge: Intel Corporation 82945G/GZ/P/PL Memory Controller Hub (rev 02) 00:02.0 VGA compatible controller: Intel Corporation 82945G/GZ Integrated Graphics Controller (rev 02) 00:1e.0 PCI bridge: Intel Corporation 82801 PCI Bridge (rev e1) 00:1f.0 ISA bridge: Intel Corporation 82801GB/GR (ICH7 Family) LPC Interface Bridge (rev 01) 00:1f.1 IDE interface: Intel Corporation 82801G (ICH7 Family) IDE Controller (rev 01) 00:1f.2 IDE interface: Intel Corporation 82801GB/GR/GH (ICH7 Family) SATA IDE Controller (rev 01) 00:1f.3 SMBus: Intel Corporation 82801G (ICH7 Family) SMBus Controller (rev 01) 01:00.0 ISDN controller: Cologne Chip Designs GmbH ISDN network Controller [HFC-4S] (rev 01) 01:02.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface 01:07.0 Ethernet controller: Realtek Semiconductor Co., Ltd. RTL-8139/8139C/8139C+ (rev 10) # cat /proc/interrupts CPU0 0:4262013 XT-PIC timer 1: 8 XT-PIC i8042 2: 0 XT-PIC cascade 3:4217220 XT-PIC qozap 5: 11979 XT-PIC eth0 8: 1 XT-PIC rtc 11: 29016 XT-PIC libata 15:4211433 XT-PIC wctdm NMI: 0 ERR: 0 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] CallerID presentation
Hi, My * (latest stable CVS) is not sending the caller id on its zap channels (digium TDM40B). The callerid is shown on call-waiting, but is hidden if the ringing channel is not already in a call. The same * and configuration was working before upgrading to the latest stable CVS. Of course I have (in zapata.conf): usecallerid = yes callwaitingcallerid = yes With defaultzone=gr (zaptel.conf) I was getting: "Didn't finish Caller-ID spill.". I changed defaultzone=us and I also tried with DEFAULT_CIDRINGS=2 (in chan_zap.c). With this setup the error is gone, but the caller id is still not sent to the phones. As I said, callwaitingcallerid is working right. Any ideas? Regards, Costa ___ 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] Re: [Asterisk-Users] Dual Xeon
> Also I would turn off Hyperthreading (in the bios). It > may cause problems. What problems? Are these digium H/W specific, asterisk specific or generaly Linux problems? Costa ___ 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] Re: [Asterisk-Users] SIP config documentation
Hi again, > Feel we need to document various solutions here. Yes we do. I still have a filling that I don't exactly get it... > Sorry, I was on the wrong topic, canreinvite has > yes|no|update as keywords. > with UPDATE a SIP method UPDATE is initiatied to change the media path. > with YES, a new INVITE is issued within the current call. (a "re-invite") > with NO, the call stays within asterisk. Sorry for asking this, but in practice, what does these mean? (yes I know I should look at the FRC... :) > I'll add comments to the bug reports. We really need to think about how > we steer the SIP channel forward. Asterisk is a multi-protocol PBX, so > there's no sense in making the SIP channel a stand-alone SIP proxy that > doesn't work with the rest of the PBX. If you don't want a PBX in the core > of your SIP network, use a SIP proxy there and use Asterisk where it fits > in. This way of reasoning means there are things that will never happen in > the Asterisk SIP channel because of the multi-protocol architecture. We > need > to be clear on that while moving the channel forward. Having said that, > there's a lot of things to do to make the SIP client and server within > Asterisk more compliant and functional with the rest of the SIP world. Olle, I think it is a mater of prioritization. If you put the switched world in the first priority, yes we should keep numbers. If you put SIP first, then we should use e-mail like addresses. After some time there might be another protocol, say XYZ. If you put XYZ first, then you should choose the method this protocol uses, etc. I believe there are three possible paths for asterisk: 1. Stick to the switched world (as the common denominator for telephony). This means that * can have any number of gateways on it, but always, it will be a "switching-like" PBX with some VoIP functionality, build in software. Example: forget about SIP, H.323 and the like, focus on switched telephony with the ability to place and receive calls via VoIP making them as switching-friendly as possible, of course with limitations. 2. Open up the possible scenarios and let the administrator choose the primary protocol. This means that although asterisk will not address in its core artificially intelligent thinks such as callerid convertion, it could be configured to follow either schema. In this case, the administrator should have the means to configure some aspects of the secondary protocols. Example: Support a number of protocols as primary and configure asterisk according to them. This means that various building blocks (like callerid handling) will be multiplied, one variation per primary protocol and the whole system will support only one protocol for such functions. When there are interfaces with other protocols the administrator should make hard decisions about the properties that cannot be converted automatically. This is my case today. I want SIP as primary and I know I will have to provide numeric callerids in the configuration when interfacing with other protocols. What I need is a SIP PBX and IVR, not a SIP proxy. I cannot do what I need with SER (at least not that easy...). 3. Build asterisk as a superset of all protocols, internally. In our case this could mean that the callerid could be defined as: callerid=TEXT or even: callerid=Your Name or just provide directory services for callerid convertions between non compatible protocols. As a superset of all protocols, asterisk will be able to be a fully functional member of each of the supported worlds and will be able to also handle all the protocols as primary at the same time. Path 3 is the perfect path. Path 2 is a good one. Path 1 demotes asterisk. If it is going to be Path 1 I believe we, all, are going to use asterisk as a secondary component in our telephony infrastructures that will provide some valuable services, but it will never be the heart of it, unless all that we need is a plain old switched-like PBX with some VoIP functionality. > We need to work together to handle the multi-domain scenario. Please > send me whatever you have and let's continue discussing this so we get > a solid architecture. I do want Asterisk to be in the forefront in > preventing > use of Asterisk as a open SIP spam relay to use mail terms. Mail servers > are > picky of which domains they server for inbound and outbound messages, in > some > cases also on what domain is used for outbound messages. We need to have > configuration that follows this line of thinking for SIP. If someone is > using > our domain for outbound calls - authenticate. If someone is randomly > placing > calls to extensions in any domain they invent *into* our PBX, don't > answer. I agree with you, but still I think all these should be configuration decisions, not implementation ones. Regards, Costa ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSU
[Asterisk-Users] Re: [Asterisk-Users] SIP config documentation
Hi all, (oej, I have lost your e-mail somehow, so I have replied to some other reply to you... sorry!) >> context= ; UP, the context name for placing calls >> >> Q1: Why is there a context for peers? > > We use peers in some other situations as well. This is strange and > rather undocumented, but an incoming call is first matched by > username with the defined users (including 'friends'). After that, > we match on IP address with the peer table. Hopefully those > peers have authentication (a secret), so we authenticate and accept > the call based on IP. This is used when connecting Asterisk as > a PSTN-gateway to other SIP proxys. Incoming calls are placed > in the peer context. Does this mean that if I have SER for example as the primary proxy for a domain and I want to use asterisk as a media gateway, I can have SER redirect my user agents to asterisk, which will be authenticated as nomral by asterisk (sip.conf entries for each agent), without using autocreatepeers=yes? >> canreinvite= ; UP yes, no or update. >> >> Q2: What the "update" option does? > > I think the update is not a keyword, but a value. Yes, my fault. > See here > http://www.voip-info.org/wiki-Asterisk%20sip%20qualify It is not a value of qualify. It is a value of canreinvite. Do you know what it does? >> callerid= ; U- caller id of the user: "Name ". > > Have to check this one. Been working a bit on this problem in the > chan_sip2 channel. I have submitted two bug reports. One includes a patch to chan_sip.c that fixes the problem. See: http://bugs.digium.com/bug_view_page.php?bug_id=0001074 Another is about CALLERIDNUM. This variable seems to strip the dots from the domain without practical reason (also SetCIDNum and the like do this). See: http://bugs.digium.com/bug_view_page.php?bug_id=0001075 >> callgroup= ; UP >> pckupgroup= ; UP >> >> Q4: Since a user cannot accept calls, why to setup call pickup for >> him/her? > Sorry, haven't used or checked call groups. Anyone else? No answer on this yet... >> language= ; U- language for voice messages and indications >> >> Q5: Why a peer does not have a language? What if we want to call >> someone >> with an IVR menu (via a .call file)? How we can choose the language >> for >> him/her? (yes, I know this can be set in the context the call will >> enter, but I think the elegant solution is to have this information >> here). >> > This is a bug fixed in the chan_sip2 channel. ok >> accountcode= ; U- CDR's account code >> incominglimit= ; U- concurrent call limitations ( >= 0 ) >> outgoinglimit= ; U- concurrent call limitations ( >= 0 ) >> >> Q6: How is it possible for a type=user phone to have BOTH incoming and >> outgoing limits? > Interesting question. Anyone else? No help on this either so far. >> nat= ; -P yes, no : Support NAT. (breaks RFC) > Well, yes, it breaks the RFC, but makes SIP work. What nat=yes really > does > is that it ignores the IP data within the registration or invite and use > the IP address Asterisk received the packet from. This works if the > client is contacting us directly, with no outbound proxy in between. > This is rather common in SIP proxy implementations right now. When STUN > and UPNP and other NAT/VoIP solutions are more frequently implemented, > the data sent to thte SIP server will not include any private NAT > networks > any more, but that will not happen overnight. ok >> fromdomain= ; -P Domain to show in the domain field of the outgoing >> call TO the peer. > I think this is mainly used when we REGISTER with an outbound proxy, > like FWD. >> fromuser= ; -P User to show in the user field of the outgoing call TO >> the peer. > Same here. fromdomain and fromuser overwrite the callerid sent TO the peer. The [general] section fromdomain and callerid are the same respectivelly (callerid in [general] is fromuser actually) for destinations not defined as peers in sip.conf (i.e. DNS SRV based lookups). The patch I have sumbitted above allows the type=user entities to have a fromdomain/fromuser given in their callerid: callerid = My Name <[EMAIL PROTECTED]> This allows asterisk serve multiple SIP domains concurrently (I am working on this for some time - I have build a configurator that automatically creates multi-domain configurations for using * for SIP virtual PBX services - If you are interested for this builder, just send me a note to send it to you - it is alpha currently). >> mask= ; -P netmask for host= parameter. > This has to be defined *before* thet host= parameter. Thanks for the hint. I didn't notice this. > What it does? Don't know. Anyone else? Why do Asterisk apply a host mask > to an IP address for a host? This is still open too. >> defaultip= ; -P if the peer does not register with us, where we should >> try to find it by default. > Useful if you restart Asterisk. The SIP device think it's registred for > a while longer, but Asterisk lost contact with them. ok. > host= > port= > means you don't use SIP REGISTER, the SIP
[Asterisk-Users] SIP config documentation
Hi all, I was trying to figure out all the valid options for a sip.conf and I believe I found a few weird things (or just a few things that are weird to me :) Anyway, I decided to post this here together with my questions and notes in case other people need this info too or have similar questions. First, let me state what the Handbook says: type=user is a SIP user agent that initiates calls (through asterisk) while type=peer is a SIP user agent that can only receive calls from asterisk. Having said that, my examination of chan_sip.c shows that: > UP < options of U=user, P=peer type= ; UP, user, peer or friend (=user+peer) context=; UP, the context name for placing calls Q1: Why is there a context for peers? permit= ; UP access control deny= ; UP access control auth= ; UP plaintext, md5 secret= ; UP user's password in plaintext md5secret= ; UP user's password in MD5 format in ":asterisk:" format dtmfmode= ; UP DTMF detection: inband, rfc2833, or info canreinvite=; UP yes, no or update. If set to no, * will be a media gateway. Q2: What the "update" option does? callerid= ; U- caller id of the user: "Name ". Q3: A lot of issues here: a. If I place here a SIP address (<[EMAIL PROTECTED]>), asterisk STRIPS the dots (it actually strips: "()-." - this is done in callerid.c) which of course destroys the callerid. Stripping of the dot character, really destroys the use of variables in contexts. Since the outgoing call from, and the incoming call to, the same user have different caller ids, a lot of asterisk's beauty is lost when programming advanced things in extensions.conf. b. If the callerid contains anything other than just a number, then when asterisk sends the call, for example, to another asterisk box (via SRV lookups) it removes the non-numeric callerid and places there the word "asterisk". This destroys the whole idea of SIP. If I place a call from the * at my office, to the * at my home, here is what is happening: Office * callerid: -> Home Phone gets: Costa Tsaousis <[EMAIL PROTECTED]> -> Costa Tsaousis Costa Tsaousis <[EMAIL PROTECTED]> -> Costa Tsaousis Costa Tsaousis <123>-> Costa Tsaousis <123> The last is AGAIN wrong, since the home phone does not know WHICH "123" it is. It should be "<[EMAIL PROTECTED]>" for the home phone to be able to call it back and for me to know which "Costa Tsaousis" called! callgroup= ; UP pckupgroup= ; UP Q4: Since a user cannot accept calls, why to setup call pickup for him/her? language= ; U- language for voice messages and indications Q5: Why a peer does not have a language? What if we want to call someone with an IVR menu (via a .call file)? How we can choose the language for him/her? (yes, I know this can be set in the context the call will enter, but I think the elegant solution is to have this information here). accountcode=; U- CDR's account code incominglimit= ; U- concurrent call limitations ( >= 0 ) outgoinglimit= ; U- concurrent call limitations ( >= 0 ) Q6: How is it possible for a type=user phone to have BOTH incoming and outgoing limits? amaflags= ; U- default, ommit, billing or documentation disallow= ; UP Disallow codecs allow= ; UP Allow codecs in order of preference insecure= ; UP yes, no : don't check source IP/port restrictcid=; U- yes, no : don't send caller id nat=; -P yes, no : Support NAT. (breaks RFC) fromdomain= ; -P Domain to show in the domain field of the outgoing call TO the peer. fromuser= ; -P User to show in the user field of the outgoing call TO the peer. host= ; -P dynamic or an IP address mask= ; -P netmask for host= parameter. port= ; -P port for host= parameter. defaultip= ; -P if the peer does not register with us, where we should try to find it by default. Q7: I am really lost with these. I understand defaultip pretty well, but then, what is exactly the use of host, port and mask, for peers? Does this have to do something with nat=yes in order to set our asterisk "public view", or what? username= ; -P username to send to the peer when calling the peer Q7: Since this is a peer option, it is really very badly documented in the various documents. All these documents state that this is used when the phone's login name is different from the default. But then, since this is a peer option it is ignored for type=user agents and is used only when asterisk is calling the phone. qualify=; -P yes, no or milliseconds : peer monitoring mailbox=; -P mailbox number for message waiting indication. These are all the options I found for peers and users
[Asterisk-Users] CAPI noise and scratches - strange behavior
Hi all, Here are some additional info & findings: My system is dual PIII 933MHz, with a Gigabyte GA-6VXDC7 mobo. Asterisk produces noises and scratches when the system is IDLE and *BOTH* CAPI channels are active (with one CAPI channel everything is fine always). When the system is busy (CPUs at 100%), sound is perfect with *BOTH* CAPI channels active. Here is how I can reproduce the problem: If I keep one of my CPUs busy while both CAPI channels are active, the noises stop. Just a scratch here and there. I keep the CPU busy with this: # while [ 1 = 1 ]; do echo " " >/dev/null; done If I run this twice (to keep busy both CPUs), then the noises appear just every a few seconds. I can reproduce this every single time. Note that I have a TDM40B (4xFXS) Digium card installed that is supposed to provide a high resolution timer to asterisk. The effect is the same with this card enabled and disabled. I know the easy workaround: Run setiathome permanently :-) However, it seems that chan_capi has some timing problems... Any help? Costa On Τετ, 2004-02-11 at 14:08, Costa Tsaousis wrote: > Hi, > > When I have two concurrent CAPI calls, * produces a lot of noises and > scratches on both CAPI channels. > > I am using SIP phones; it appears on all phones, even if two separate SIP > devices are connected to the two CAPI channels. > > The problem does not appear with any number of concurrent calls using SIP > end-to-end (with * as a media gateway). > > I am using FritzCard! DSL (the ISDN part of it) and kernel 2.4.24 vanilla. > > Any help is appreciated. > > Costa > ___ 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] High Density configuration for Voice & Fax
Hi, Are there any well known good H/W configurations for high density E1 setups supporting * and FAX? The response we got from digium about their cards, as far as FAX is concerned, is: "I would not depend upon them for FAX. It does work, but it is not completely reliable.". The minimum we would need is 4 x E1 on each system but both Voice and FAX at the same time. Also, I saw somewhere on the asterisk sites, that * can support up to 100 concurrent calls on a dual Xeon 1.8GHz when doing media conversion. We have a few projects that we will need PRI <-> GSM or G729 media conversion (for WAN use). Do you know any cards that can take care of the media conversion and free the CPUs (for use with * of course)? Kind regards, Costa ___ 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] Noise and scratches when there are two concurrent CAPI calls
Hi, When I have two concurrent CAPI calls, * produces a lot of noises and scratches on both CAPI channels. I am using SIP phones; it appears on all phones, even if two separate SIP devices are connected to the two CAPI channels. The problem does not appear with any number of concurrent calls using SIP end-to-end (with * as a media gateway). I am using FritzCard! DSL (the ISDN part of it) and kernel 2.4.24 vanilla. Any help is appreciated. Costa ___ 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
