[asterisk-users] Asterisk stops processing SIP UDP messages

Last week I had a couple of outages one machine, the problem was that Asterisk suddly stopped responding to UDP SIP requests. tcpdump show requests arriving on the machine, sip debug log in asterisk doesn't show anything for the UDP peers, TCP functions just fine. In all 3 cases the log is

[asterisk-users] Debug compile fails

DEBUG_CHANNEL_LOCKS MALLOC_DEBUG $ make make install $ asterisk asterisk -rx core show locks No such command 'core show locks' (type 'core show help core' for other possible commands) Am I missing something? -- Daniel Tryba

Re: [asterisk-users] Debug compile fails

On Fri, Sep 24, 2010 at 02:30:12PM -0400, Paul Belanger wrote: Am I missing something? DEBUG_THREADS Thanks, I guess I should have RTFM :) -- Daniel Tryba -- _ -- Bandwidth and Colocation Provided by http://www.api

Re: [asterisk-users] Problems compiling Asterisk on Debian

to continue [Y/n]? -- Daniel Tryba -- _ -- 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

Re: [asterisk-users] Problems compiling Asterisk on Debian

are running stable you should either use the backports version: http://packages.debian.org/lenny-backports/dahdi Or make your own package from testing/unstable sources. -- Daniel Tryba -- _ -- Bandwidth and Colocation

Re: [asterisk-users] How to pick a codec on the fly

; Need it the other way so I can do DAHDI-- IAX testing. exten = 1234,1,Set(_SIP_CODEC=alaw) exten = 1234,n,Goto(0234,1) exten = 2234,1,Set(_SIP_CODEC=slin) exten = 2234,n,Goto(0234,1) Should do the trick. -- Daniel Tryba

Re: [asterisk-users] NAT issue (i think?)

Enable). But even with both these setting enabled NAT gateways sometimes seem to lose track of SIP sessions (I have more trouble with Cisco devices than Linux routers), setting the UDP session timeout to 10m seems to help (default is something like 3m). -- Daniel Tryba

Re: [asterisk-users] NAT issue (i think?)

device and Asterisk. -- Daniel Tryba -- _ -- 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

[asterisk-users] AMI getting related channels in Ringing state

', 'CallerIDNum' = 'trunk', 'CallerIDName' = '0031234567890', 'Accountcode' = '', 'ChannelState' = '4', 'ChannelStateDesc' = 'Ring', 'Context' = 'macro-dial-one', 'Extension' = 's', 'Priority' = '37', 'Seconds' = '3', 'Uniqueid' = '1286364290.8474', ) -- Daniel Tryba

Re: [asterisk-users] AMI getting related channels in Ringing state

On Wed, Oct 06, 2010 at 01:56:55PM +0200, Daniel Tryba wrote: Issuing the AMI Status command results in a list of active channels. But how to figure out which channels are related before the call is answered? CoreShowChannels gives a little bit of extra data in the originator channel

Re: [asterisk-users] 401 Unauthorized with Snom but not with Zoiper softphone

to 'yes' or something else? -- Daniel Tryba -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http

Re: [asterisk-users] 401 Unauthorized with Snom but not with Zoiper softphone

On Thu, Oct 07, 2010 at 02:24:59PM +0200, Daniel Tryba wrote: It's the same account, the same password, but other agent. Can anyone help me with this please ?! I see no difference but there must be !! The difference is the SNOM is using rport and Zoiper isn't. Is nat for this client

Re: [asterisk-users] AMI getting related channels in Ringing state

On Wed, Oct 06, 2010 at 01:56:55PM +0200, Daniel Tryba wrote: Issuing the AMI Status command results in a list of active channels. But how to figure out which channels are related before the call is answered? Anybody? My workaround for this problem is setting a persistent variable

Re: [asterisk-users] 401 Unauthorized with Snom but not with Zoiper softphone

the difference values for nat to see if one works with the NAT gateway or use STUN like suggested elsewhere. -- Daniel Tryba -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us

Re: [asterisk-users] Voice quality assessment in Asterisk

be retrieved from the RTCP reports in asterisk). These probes and the analyzer software aren't bug free and perfect but give a good indication of all historic calls. Once a problem is spotted we move to test calls to trace the problem. -- Daniel Tryba

Re: [asterisk-users] Voice quality assessment in Asterisk

. -- Daniel Tryba -- _ -- 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

Re: [asterisk-users] Routers that do not show external IPs...

ports did you open? Only sip or also the RTP ports? -- Daniel Tryba -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs

Re: [asterisk-users] SIP - no audio behind nat problem

to an * (behind NAT) without problems and without any portforwards at the Grandstream side. nat=yes and canreinvite=no as always do the trick for me. -- Daniel Tryba -- _ -- Bandwidth and Colocation Provided by http://www.api

Re: [asterisk-users] Asterisk to switch on electric heaters remotely?

this is a the combination of 2 examples (CIRC10 and CIRC11 from http://ardx.org/src//guide/2/ARDX-EG-OOML-DD.pdf). -- Daniel Tryba -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us

Re: [asterisk-users] Same extension registering over eth0 and eth1

stateless. There where a duplicate TCP packet will be ignored, UDP will be processed twice. While I didn't have this problem with SIP/RTP in a hot failover network as described, we changed a hot setup to cold standby because of troubles with UDP traffic (can't remember the specifics). -- Daniel

Re: [asterisk-users] scratchy sound on TE410P

, and the audio is overdriven. It could be the echo canceller, I had this kind of problem with OSLEC. I also thought the PRI provider was sending clipped audio. I switched to the VPM450 daughterboard and since audio has been crystal clear. What is your setup for echo cancelling? -- Daniel

Re: [asterisk-users] scratchy sound on TE410P

problems dissapear but overall voice quality is lower IMHO. You could try disabling ec all together and check if clipping still occurs. But it does sound like an operator problem if you get errors. -- Daniel Tryba

Re: [asterisk-users] ISDN-FAX with Asterisk

to the sender). -- Daniel Tryba -- _ -- 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

Re: [asterisk-users] call forward problem

${DB(CFIM/${EXTEN})} is empty (and do nothing) or set (and dial that number instead). -- Daniel Tryba -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory

Re: [asterisk-users] Asterisk pass a call to status answer while still ringing

hangup before someone really answer, the call is reported as answered but it isn't. This gives me problem for call charge. Some I idea what can be? It is the Early Connect (early-connect) setting on the SIP interface. If it is enabled the call will be answered in my SN4554. -- Daniel

Re: [asterisk-users] Someone has hacked into our system

) of your internet connection(s), by only explicitly listing internal ipadresses and hostnames. e.g.: domain=10.2.3.4 domain=sip.example.com The standard scanners will get a Not a local domain error, since they only try the external ipadress to connect (for now). -- Daniel Tryba

Re: [asterisk-users] DAHDI on VMWARE

that should do the trick if you want something internal)). -- Daniel Tryba -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs

Re: [asterisk-users] Execute DialPlan Context without Answer App

= _X.,2,Set(ALLARME=${EXTEN:1:1}) exten = _X.,3,AGI(checkgroup.php|${GRUPPO}) -- Daniel Tryba -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory

Re: [asterisk-users] Asterisk SIP attacks and sshguard

is more familiar with the attacks or how to generate these messages would give me some assistance, or chime in on the sshguard-users list, that'd be most appreciated. You could use SIPVicious to run attacks on your own servers: http://code.google.com/p/sipvicious/ -- Daniel Tryba

[asterisk-users] Unexpected dialplan match

)}) [pbx_config] 3. Set(extension=${CUT(EXTEN,*,3)}) [pbx_config] 4. Set(CDR(accountcode)=${accountcode}) [pbx_config] 7. ResetCDR() [pbx_config] 8. ... -- Daniel Tryba

Re: [asterisk-users] Unexpected dialplan match

for is a extension that handles: ^\*\w+\*\d+$ I guess I'll have to catch _*. and manually check if it matches above regexp. -- Daniel Tryba -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join

Re: [asterisk-users] app_voicemail.c how to enable debugging?

is unavail.gsm exists from the dialplan to add 's' to the arguments. Implementing this in an AGI script should be trivial. -- Daniel Tryba -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk

Re: [asterisk-users] Forward voicemail to group of people

the dial plan but that is not available when you are listening to voicemail. My guess the easiest hack is to create a local alias (/etc/aliases) that will relay the mail to multiple users. -- Daniel Tryba -- _ -- Bandwidth

Re: [asterisk-users] sip attack.. fail2ban not stopping attack

On Sat, Dec 25, 2010 at 04:04:59PM -0700, Dave George wrote: My server is being attached all day and fail2ban is not stopping the attack. I updated stamstamp to match fail2ban requirements. How about posting your fail2ban config? -- Daniel Tryba

Re: [asterisk-users] live audio stream in asterisk

. But if you take a look at the content of the .asx you'll see that it contains links to mp3 streams. You could pick one of them manually, but expect the URLs to change in the future and break your MOH. -- Daniel Tryba

Re: [asterisk-users] sip attack.. fail2ban not stopping attack

. Daniel Tryba -- _ -- 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

[asterisk-users] Queues, priorities and (miscalculated) holdtimes

and waittime should avoid the wrong reported holdtime. -- Daniel Tryba -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs

Re: [asterisk-users] Force different codecs on call base

://issues.asterisk.org/view.php?id=13243 explains settings codecs (and its difficulties, but since you have full control there shouldn't be a problem). Use GROUP and GROUP_COUNT to find out how many channels are are active, use this to decide whether to use alaw or g726. -- Daniel Tryba

Re: [asterisk-users] how to get Current Calls details

by matching channel==dstchannel for all channels. -- Daniel Tryba -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http

Re: [asterisk-users] Email alerts for trunks (peers)

changes: http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+Events -- Daniel Tryba -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar

Re: [asterisk-users] Dial command

scripting language. -- Daniel Tryba -- _ -- 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

Re: [asterisk-users] Carrying context from one server to another?

transport. I use this to do multitenant billing on the remote server in places where I only want 1 IAX trunk. Whether this is effective depends on your control of the local server. -- Daniel Tryba -- _ -- Bandwidth

Re: [asterisk-users] Two Asterisk machines for redundancy

still need more than 1 if you want to avoid this being a single point of failure. But they can be more flexible in some setups (multiple active asterisk machines connecting simulataniously) -- Daniel Tryba -- _ -- Bandwidth

Re: [asterisk-users] Using voice modem as poor man's FXO in Asterisk 1.8

). Thus, a 10/100Mbps Ethernet card was installed to provide the second port needed. You can free the PCI slot if you use VLANs on the internal interface to seperate the internal and external traffic. This requires a switch with vlan support, shouldn't could much more than $80. -- Daniel Tryba

Re: [asterisk-users] Using voice modem as poor man's FXO in Asterisk 1.8

. -- Daniel Tryba -- _ -- 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

[asterisk-users] QSIG-SIP overlap dialing and Asterisk (RFC4497)

requested. -- Daniel Tryba -- _ -- 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

Re: [asterisk-users] Prompt for PIN After dialing

(accountcode)} ]?dial:fail) exten = _X.,n(dial),Dial(SIP/${EXTEN}@trunk) exten = _X.,n,Hangup() exten = _X.,n(fail),Playback(notauthorized) exten = _X.,n,Congestion() Add prompts and maybe Answer where needed. -- Daniel Tryba

Re: [asterisk-users] Good, inexpensive wireless VOIP handsets ?

, reasonable voice quality), but an ATA has a better price point (Linksys PAP2T with 2 cheap handsets). So far all SIP DECT solutions I tried suck on some level. -- Daniel Tryba -- _ -- Bandwidth and Colocation Provided

Re: [asterisk-users] how to add-edit-delete entery into asterisk conf files

Action-00:append Cat-00:newuser Var-00:secret Value-00:nottelling -- Daniel Tryba -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory

Re: [asterisk-users] Beggining asterisk

about Dahdi. Replace zap(tel) with dahdi. But I suggest starting with the Ubuntu bundeled packages (asterisk/dahdi/libpri). -- Daniel Tryba -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New

Re: [asterisk-users] Asterisk on Android?

of the phone (RIL) at this moment. So if you want to use the GSM itself you are out of luck. -- Daniel Tryba -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory

Re: [asterisk-users] pickup for extension in asterisk 1.4?

,?the following error appears?in the CLI: Where are you *setting* the PICKUPMARK? See the examples on http://www.voip-info.org/wiki/view/Asterisk+cmd+Pickup -- Daniel Tryba -- _ -- Bandwidth and Colocation Provided by http

Re: [asterisk-users] ISDN2 PCIe Card for Asterisk

(once you figure out how with a SmartNode), good quality and much easier in failover (though might be a single point of failure by itself). -- Daniel Tryba -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com

Re: [asterisk-users] How does AMI work with events ?

weird hangups in PHP with longlived sockets. Personally I monitor SIP/IAX peers by parsing: asterisk -nrx sip show peers with Nagios/NRPE scripts. -- Daniel Tryba -- _ -- Bandwidth and Colocation Provided by http://www.api

Re: [asterisk-users] pick up code

On Tue, Sep 06, 2011 at 04:43:39PM +, salaheddine elharit wrote: [asterisk 1.4] [agents] exten = _2XX,1,Dial(SIP/${EXTEN}) exten = _*8XXX,1,PickupChan(SIP/${EXTEN:2}) SIP/222 is not a channel but an extension. See: http://www.voip-info.org/wiki/view/Asterisk+cmd+Pickup -- Daniel

[asterisk-users] Overlap SIP dialing

Looking at the history of the list I don't expect any answer but lets try anyway: Does anybody use overlap dialing from SIP devices to asterisk? Does anybody have a working example? -- Daniel Tryba -- _ -- Bandwidth

Re: [asterisk-users] Overlap SIP dialing

. -- Daniel Tryba -- _ -- 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

Re: [asterisk-users] Overlap SIP dialing

(was only concerned about billsec). As far as I can see this doesn't happen in 1.6.2.x. The lack of destination in CDR makes overlap dialing useless since I can't bill my customers. -- Daniel Tryba -- _ -- Bandwidth

Re: [asterisk-users] Overlap SIP dialing

it into account. -- Daniel Tryba -- _ -- 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

Re: [asterisk-users] Digium ISDN card

/products/digital/single-span/ Exact model depends on type of PCI interface. Don't forget to set the jumper from T1 to E1 :) -- Daniel Tryba -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk

Re: [asterisk-users] DTMF problem

softphone you are sending out-of-band DTMF which is basically SIP messages. You can emulate this feature from the Expensive PBX system by setting: relaxdtmf=yes in the case of SIP, option may vary with Techology. -- Daniel Tryba

Re: [asterisk-users] DID and how the caller id will appear

on PRIs is your whole number excluding the national prefix. In your case Set(CALLERID(num)=65631040) You might try prefixing national TON and e164 format options: Set(CALLERID(num)=Ne65631040) -- Daniel Tryba

Re: [asterisk-users] [asterik-users] Installing PRI card

modinfo)). -- Daniel Tryba -- _ -- 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

Re: [asterisk-users] Asterisk in the Cloud with Diamonds

On Sun, Oct 02, 2011 at 12:05:54PM -0400, Nick Khamis wrote: I was hoping to get some your experiences regarding putting asterisk on a cloud. A first obvious limitation is the number of ports used by asterisk to transfer voice packets. So what obvious limitation am I missing? I always

Re: [asterisk-users] Maybe slightly OT but..

provider and VOIP provider is that creating a callback or originator setup is still cheaper then using the GSM carrier (esp. long distance), has better quality (lower latency) and still works if data is not available. -- Daniel Tryba

Re: [asterisk-users] CallerID inconsistently presented through ISDN/cellular networks

the problem and correct a misconfigured switch. It also helps to be able to route all mobile traffic through an other provider, if they start to lose lots of minutes providers will act. -- Daniel Tryba -- _ -- Bandwidth

Re: [asterisk-users] Emulate and script emulation of users calling in/receiving calls, transferring calls etc

to/from Echo(), voicemail or IVRs! But much easier is to have multiple (soft)phones available to 1 student. -- Daniel Tryba -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us

Re: [asterisk-users] Extenxions Optimization

On Sun, Jun 09, 2013 at 10:30:45AM +0200, Olivier CALVANO wrote: We want optimize my extensions file conf on asterisk 11.4.0 : ; Destination: Gambia Type: Fixe exten = _00220X.,1,Set(CDR(CodeCom)=BUS-GMB) [5 lines] ; Destination: Libya Type: Fixe exten =

Re: [asterisk-users] I need a second opinion on a new phone system deployment

On Fri, Jun 14, 2013 at 09:43:29AM -0600, Nunya Biznatch wrote: System will use G.722 for VoIP Phones. [...] 2-servers acting as gateways. Each handling 2 PRIs for outside trunks. So why use g722? Just use your local g711 law and thus avoid the transcoding impact to/from the PSTN and calls

Re: [asterisk-users] Issue dialing out

Jun 15 13:06:05 VERBOSE[30232]: -- Executing Dial(Zap/1-1, zap/g1/1XX|20|tT) in new stack Jun 15 13:06:05 VERBOSE[30232]: -- Called g1/1XX Jun 15 13:06:08 VERBOSE[30232]: -- Zap/2-1 answered Zap/1-1 Jun 15 13:06:08 VERBOSE[30232]: -- Attempting native bridge of Zap/1-1 and

Re: [asterisk-users] I need a second opinion on a new phone system deployment

On Sat, Jun 15, 2013 at 10:28:50AM -0600, Nunya Biznatch wrote: Answer - There's a couple reasons I'm thinking this way, which may be misguided so thanks for making me think about it. First is redundancy. Offloading the PRIs and analog phones from the primary PBX means if there's an issue, I

Re: [asterisk-users] Issue dialing out

On Sat, Jun 15, 2013 at 03:02:41PM -0400, Andre Goree wrote: Setting the CID did not work, unfortunately :( [...] I'm going to try another number that we have through them in hopes that it'll complete and I'll let you know if that works. Do you have any other suggestions on what you think

Re: [asterisk-users] Issue dialing out

On Sat, Jun 15, 2013 at 04:24:21PM -0400, Andre Goree wrote: Thanks so much for your suggestions. I'm running 1.0.x (yes, archaic, and in fact my actual task is migrating this system to asterisk11+Freepbx -- very fun in and of itself without regards to this issue...but I digress), and so I

[asterisk-users] Getting source ip adress of incoming INVITE

I'm interested in finding out what the source ip is of an invite in the dialplan (Asterisk 11). As far as I can see this information isn't accessible. The only solution I can think of is parsing either Record-Route or Via headers. This is for recognizing guests in the default context for sip. --

Re: [asterisk-users] Getting source ip adress of incoming INVITE

On Fri, Jul 04, 2014 at 10:04:45AM -0400, Richard Kenner wrote: I'm interested in finding out what the source ip is of an invite in the dialplan (Asterisk 11). ${CHANNEL(recvip)} Doh! That is an obviouls place to look. I'm wondering why I didn't think about this or couldn't find any hints.

[asterisk-users] PJSIP with registratrion to DNS SRV records fail with PJLIB_UTIL_EDNSNOANSWERREC

I finally thought it might be a good time to start looking at the pjsip implementation in Asterisk 13. But trying to register to a sip cluster that uses SRV records fails randomly with: [Nov 4 15:50:59] WARNING[31330]: pjsip:0 : tsx0x7f075c006 Failed to send Request msg REGISTER/cseq=17800

Re: [asterisk-users] Pass CallerId/Privacy info from A Leg to B Leg

On Thu, Aug 17, 2017 at 07:28:00AM +, Grant Bagdasarian wrote: > Is there an option to give to the Dial command, or another variable to set, > to make Asterisk copy such information to the B Leg? > Or do I have to program this out myself? In chan_sip there are the trustrpid and sendrpid

Re: [asterisk-users] Support for inbound UPDATE request

On Fri, Jul 07, 2017 at 07:44:26PM +0530, Rahul MathuR wrote: > Could you please let me know whether the latest Asterisk has a support for > inbound UPDATE ? > > In my case, the carrier is sending an UPDATE to change the codec which is > replied by 5xx from Asterisk 11.17.1. Asterisk 13/PJSIP

Re: [asterisk-users] DMTF payload bug in 13.14.1 with pjsip and direct_media?

On Thu, Jun 29, 2017 at 11:55:51AM -0500, Richard Mudgett wrote: > > To me this looks like a bug in asterisk. Either asterisk should use the > > same rtp payloads for telephone-events on both call legs during inital > > callsetup or asterisk should come to the conclusion there is an > >

Re: [asterisk-users] PJSIP equivalent for SIPDtmfMode?

> > Can't find a way to control the dtmf mode on a per session basis with > > pjsip, used to use SIPDtmfMode from the dialplan with chan_sip. Any > > hints on how to do this? > > There is no current way, but a community member has recently posted a > change[1] for review which implements this. >

[asterisk-users] PJSIP equivalent for SIPDtmfMode?

Can't find a way to control the dtmf mode on a per session basis with pjsip, used to use SIPDtmfMode from the dialplan with chan_sip. Any hints on how to do this? -- _ -- Bandwidth and Colocation Provided by

[asterisk-users] DMTF payload bug in 13.14.1 with pjsip and direct_media?

While trying to use direct_media I'm seeing RTP payload mismatches after succesful reinvites. Initial INVITE from endpoint A to asterisk has rfc4733 DMTF m=audio 35648 RTP/AVP 9 8 111 96 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 >From asterisk to upstream U: m=audio 14338 RTP/AVP 9 8 111

Re: [asterisk-users] pjsip direct_media=yes and "unknown" endpoints

> > Anybody got an idea why the last scenario fails to work? > > If you turn up core debug (core set debug 2) and ensure it is going to > the CLI then the bridge_native_rtp module will tell you why exactly it > can't native bridge. You might also want to do a core show channel on > both channels

[asterisk-users] pjsip direct_media=yes and "unknown" endpoints

I'm trying to implement direct_media between multiple peers and an uplink provider, all of whom have direct_media=yes configures. For originating calls to the uplink provider direct_media=yes works like expected. SIP flows through asterisk, rtp doesn't SIP: enduser <-> SBC <-> asterisk 13 <->

Re: [asterisk-users] Change OS from CentOS 6 to 7

On Fri, Aug 04, 2017 at 03:27:40PM -0400, Jerry Geis wrote: > Audio packets are running... > > 961 16.150421076 192.168.5.150 -> 192.168.5.25 RTP 214 PT=ITU-T G.711 PCMU, > SSRC=0x6A3E0AF1, Seq=28402, Time=73280 > 962 16.170411284 192.168.5.150 -> 192.168.5.25 RTP 214 PT=ITU-T G.711 PCMU, >

Re: [asterisk-users] pjsip direct_media=yes and "unknown" endpoints

On Wed, Apr 26, 2017 at 06:25:43PM +0200, Daniel Tryba wrote: > Whoever when a terminating call comes in from the uplink provider, a > sip request is send to a redirector. The redirector has > redirect_method=uri_core configured (the only method that works for > me). [...] > The r

Re: [asterisk-users] German sip dial rules

On Mon, Jun 12, 2017 at 05:00:31PM +0200, Hans-Peter Jansen wrote: > is somebody attending, that wants to share his outgoing dial rules of > extension.conf, like used in typical(?) german pbx setups? > > * zero prefix for outside calls > * zero zero or plus prefix for international calls > *

Re: [asterisk-users] Writing CDR's to two database servers

On Mon, Jun 19, 2017 at 11:47:04AM -0400, Tech Support wrote: > I know that there are probably several solutions to this problem, but > what I am trying to do is provide some redundancy for my customers CDR data. > I know that doing simple backups of MySQL is probably the easiest way to go, >

Re: [asterisk-users] Let's encrypt privkey : Specified certificate file could not be used

On Fri, Jun 02, 2017 at 02:36:38PM +0200, Jonas Kellens wrote: > [Jun 2 14:29:28] == DTLS ECDH initialized (secp256r1), faster PFS enabled > [Jun 2 14:29:28] ERROR[27360][C-0ae5]: res_rtp_asterisk.c:1441 > ast_rtp_dtls_set_configuration: Specified certificate file >

[asterisk-users] OT: DMARC enabled domains on this list

Having enabled a strict DMARC setup I noticed everytime I send a message here I get all these reports of messages which fail DMARC. Since I don't want people to miss my wise thoughts maybe the maintainers of this list could look into DKIM signing (or any of the other ways to work around spf and

Re: [asterisk-users] OT: Want to capture all SIP messages

On Wed, May 31, 2017 at 01:39:25PM -0700, Steve Edwards wrote: > >What bugs you about the output format? > > It's been a while, but as I recollect, it included the date/timestamp in the > file name of the 'ring buffer' which meant that each time the host was > rebooted, dumpcap didn't know the

Re: [asterisk-users] OT: Want to capture all SIP messages

On Wed, May 31, 2017 at 12:36:47PM -0700, Steve Edwards wrote: > I want to capture all SIP messages. > > I have about 30 hosts in about 6 colos. > > My first thought was dumpcap, but the output file name format bugs me. > > What do you use for long term SIP capture? What bugs you about the

Re: [asterisk-users] Forward error code beetwen legs

On Thu, Jun 01, 2017 at 09:06:25PM +0200, Loic Chabert wrote: > [gotoexternal] > exten => _X.,1,Dial(SIP/${EXTEN}@provider) > > When my SIP provider return to asterisk a 404 SIP error code, asterisk > return to phone a 503 error code. > > Why 503 error code has been returned, and not the

Re: [asterisk-users] Extensions of sip trunk

On Tue, Jun 06, 2017 at 12:40:21AM +0200, Hans-Peter Jansen wrote: > > Yes, something like if they can't fix the R-URI: > > exten => X_.,n,Set(TO=${CUT(SIP_HEADER(To),@,1)}) > > exten => X_.,n,Set(TO=${CUT(TO,:,2)}) > > exten => X_.,n,Goto(somewhereelsetopreventloops${TO},1) > > Sorry for the

Re: [asterisk-users] OT: DMARC enabled domains on this list

On Tue, Jun 06, 2017 at 08:23:33AM -0400, James B. Byrne wrote: > > The reports are there to tell you something isn't right (like on this > > mailing list). Disabling them is only hiding the problem, people might > > be replying with the correct answer to a problem, but the OP might > > never gets

Re: [asterisk-users] asterisk server - no sound

On Tue, Jun 06, 2017 at 03:18:32PM +0200, andre castro wrote: > extensions.conf: > [home] > exten = 102,1,Answer() > same = n,Wait(1) If this is copy and paste, then your dialplan is broken (= should be =>) But to debug, enable logging (core set verbose 5), when needed debugging (core set debug

Re: [asterisk-users] pjsip user_eq_phone adds user=phone to anonymous user bug?

On Fri, Jun 09, 2017 at 11:40:01AM -0300, Joshua Colp wrote: > What seems to be happening is that the session is being set up and the > user=phone parameter added. It's only after that the values are updated > to be Anonymous and the user=phone parameter is left there. Please file > an issue[1]

[asterisk-users] pjsip user_eq_phone adds user=phone to anonymous user bug?

With pjsip (asterisk 13.14.1) I see the problem that an anonymous from header gets user=phone appendend to the URI if user_eq_phone=yes is specified: On the incoming leg: From: anonymous ;tag=Q5zBj7BMnvI6Fe6O2866fox3ZHmn-smt Get transformed to From:

Re: [asterisk-users] Asterisk 1.6.2 how to debug T.38 udptl problems

On Thu, Jun 15, 2017 at 12:11:36PM +0200, Benoit Panizzon wrote: > Or does anyone have an idea over what the asterisk is stumbling? What if you set the maxdata in asterisk to a value lower than the other side? e.g. sip.conf: t38pt_udptl = yes,fec,maxdatagram=400 --

Re: [asterisk-users] asterisk 13.16 / pjsip / t.38: res_pjsip_t38.c:207 t38_automatic_reject: Automatically rejecting T.38 request on channel 'PJSIP/91-00000007'

On Sun, Jun 11, 2017 at 01:16:10PM +0200, Michael Maier wrote: > Let's go into details: > I'm starting at the point where asterisk / fax client receives the T.38 > reininvite from the server from the provider 195.185.37.60:5060 for the > fax client (extension 91): I'm running Asterisk 11 on my

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