Re: [asterisk-users] Is there any way to pass caller id to cell phone?

On Thu, Oct 11, 2018 at 05:18:24PM +, Ivan Demkovitch wrote: > Where problem comes in - if person not at the desk - his cell phone shows > call from OFFICE number and there is no way to tell who is really calling. > We use Callcentric as a trunk if it makes any difference. > I'd like to add

Re: [asterisk-users] Disabling a trunk at runtime

On Fri, Oct 12, 2018 at 07:59:52AM -0400, Telium Support Group wrote: > I have an Asterisk system with 2 trunks (as shown below). I need to be able > to disable a trunk at runtime. I may not change the dialplan but I can > change sip.conf and reload. > > Any attempt to dial in the dialplan uses

Re: [asterisk-users] 401 unauthorized

On Wed, Aug 29, 2018 at 11:37:34AM -0400, Jerry Geis wrote: > I have a connection to a cisco all manager SIP trunk. The first call coming > across CCM to the asterisk server works fine... Then when I do a second > call from CCM to asterisk I am getting a SIP 401 unauthorized. > > My definition

Re: [asterisk-users] PJSIP redirect_method=uri_core and header modifications

On Fri, Aug 03, 2018 at 04:24:06PM +0200, Daniel Tryba wrote: > redirect_method=uri_pjsip works as expected with regard to the header > manipulation stuff. > > Also I can't remember why, in the past, I decided to not use uri_pjsip > other than having the redirected host in CDR

Re: [asterisk-users] PJSIP redirect_method=uri_core and header modifications

On Thu, Aug 02, 2018 at 05:29:23PM +0200, Daniel Tryba wrote: > With chan_sip there is the variable SIP_MAX_FORWARDS to set > Max-Forwards. This counter is persistant after a redirect. I can't find > the equivalent for PJSIP, so I went the way of header manipulation. Only > to find

Re: [asterisk-users] 400 reply to INVITE not properly treated

On Thu, Aug 02, 2018 at 02:40:48PM +1000, Patrick Wakano wrote: > In my opinion, Asterisk should at fail the Dial and proceed with whatever > was configured in the dialplan I tried some other 4XX SIP codes, but > the only one I found not behaving properly is the 400 one I think you are

Re: [asterisk-users] PJSIP redirect_method=uri_core and header modifications

On Thu, Aug 02, 2018 at 05:29:23PM +0200, Daniel Tryba wrote: > With chan_sip there is the variable SIP_MAX_FORWARDS to set > Max-Forwards. This counter is persistant after a redirect. I can't find > the equivalent for PJSIP, so I went the way of header manipulation. Only > to find

[asterisk-users] PJSIP redirect_method=uri_core and header modifications

With chan_sip there is the variable SIP_MAX_FORWARDS to set Max-Forwards. This counter is persistant after a redirect. I can't find the equivalent for PJSIP, so I went the way of header manipulation. Only to find out that any headers added to the outbound leg are lost after a redirect (with

Re: [asterisk-users] Do you set chan_sip's ignoresdpversion to true ?

On Tue, Jun 19, 2018 at 07:38:12PM +0200, Olivier wrote: > I've just discovered chan_sip's ignoresdpversion setting. > Do you use it ? > If positive which kinnd of issue could you solve with it ? IIRC I used to enable this option when talking to some Ericsson SBC. It solved a problem concerning

Re: [asterisk-users] How to ignore REFER entirely with chan_sip or PJSIP ?

On Fri, Jun 15, 2018 at 05:32:30PM +0200, Olivier wrote: > In my testing, I saw that Asterisk always included a REFER value in each > INVITE's Allow header, no matter how allowtransfer/allow_tranfer was set. > > Is there a way to remove this REFER value entirely either globally or > specifically

Re: [asterisk-users] Questions about SIP From, P-Asserted-Id fields and Diversion headers ?

On Tue, Jun 05, 2018 at 11:34:51AM +0200, Olivier wrote: > 1.According SIP RFCs, is possible/recommended to have different values in > From and P-Asserted-Id fields ? > For instance, From field showing 123456789 and P-Asserted-Id showing > 987654321 (beside privacy considerations) ? Yes, most

Re: [asterisk-users] Long extensions that contain dashes

On Tue, May 29, 2018 at 08:32:39PM -0700, David P wrote: > We would like to use 20-char extension values that use dashes and alphanums > after the first four digits. In order to handle these via pattern-matching, > how can I define a pattern that allows dashes? There seems to be no option > at

Re: [asterisk-users] Decoding SIP register hack

On Thu, May 17, 2018 at 12:27:17PM -0400, sean darcy wrote: > > WARNING.* .*: fail2ban='' > > > ># Option:  ignoreregex > ># Notes.:  regex to ignore. If this regex matches, the line is ignored. > ># Values:  TEXT > ># > >ignoreregex = > > > > > Thanks. Very useful as a tutorial for

Re: [asterisk-users] When should a Progress or Ringing be used in a today's telephony ?

On Wed, May 16, 2018 at 04:51:49PM +0200, Olivier wrote: > 1. When Asterisk receives a SIP call coming from PSTN, is there a time > frame within which Asterisk must reply something to keep caller from > canceling the call ? Where does this limit come from ? From SIP RFC ? From > local regulation

Re: [asterisk-users] Streaming MoH from iHeart radio?

On Wed, May 16, 2018 at 11:01:53AM -0400, Mike Diehl wrote: > I have a user who would like to stream their favorite radio station from > iHeart radio for their music on hold. > > It this TECHNICALLY possible? Yes. > If so, any pointers would be appreciated.

Re: [asterisk-users] SIP Codec negotiation

On Thu, May 10, 2018 at 11:44:14AM -0700, Steve Edwards wrote: > I receive an INVITE/SDP containing: > > m=audio 11310 RTP/AVP 3 0 101 > > which I interpret as gsm, ulaw, rfc2833. > > and I reply with an OK/SDP containing: > > m=audio 15884 RTP/AVP 0 3 101 > > which I interpret as

Re: [asterisk-users] multi step auth?

On Tue, May 08, 2018 at 03:04:55PM -0500, Jeff LaCoursiere wrote: > Thats till doesn't change the SIP header.?? Basically they want to send a RE > INVITE and authenticate my DID number.?? But my DID number does not have a > peer or user entry in sip.conf.?? Perhaps I am answering my own question,

Re: [asterisk-users] Pass through registration / proxy

On Wed, Apr 11, 2018 at 12:04:18PM -0400, Telium Technical Support wrote: > Maybe proxy is the wrong word I chose. Asterisk is something like a peer to > the legacy PBX. I thought about setting up individual SIP accounts on the > Asterisk box to connect to the legacy PBX, or maybe a SIP trunk to

Re: [asterisk-users] Pass through registration / proxy

On Tue, Apr 10, 2018 at 09:22:02PM -0400, Telium Technical Support wrote: > I need to create a SIP proxy to be placed in front of a legacy PBX. When a > phone registers with the proxy, I would like Asterisk to register with the > PBX behind it. (To tell the PBX to send calls to the proxy and

Re: [asterisk-users] PJSip CallerID Question

On Fri, Apr 06, 2018 at 02:27:31PM -0500, Brent Davidson wrote: > I have multiple Asterisk instances set up in different locations and would > like to modify the callerID of inbound calls to identify which instance the > call is coming from.  I knew how to do that with the old sip format, but >

Re: [asterisk-users] Setting outgoing CALLERID without changing CDR(src)

On Wed, Mar 28, 2018 at 08:16:26PM -0600, Carlos Chavez wrote: > ?? I thought I had found and answer to this question by using > CALLERID(ani) but it seems that only works on versions prior to 12.?? On > Asterisk 13 setting CALLERID(num) before dialing to an external trunk always > changes

Re: [asterisk-users] Is 100 trying mandatory? Can asterisk answer with 180 without prior 100 trying?

On Mon, Mar 19, 2018 at 12:59:47PM -0300, Joshua Colp wrote: > > To try to reproduce the problem with our SBC, is there a way to tell > > the asterisk, preferably PJSIP, to directly answer with 180 ringing > > without prior 100 trying? > > The PJSIP channel driver has no option or ability to do

Re: [asterisk-users] Avaya 9608G and DHCP and TFTP and HTTP oh my

On Wed, Mar 07, 2018 at 10:08:52PM +, Thomas Peters wrote: > You did indeed warn me. I've made progress, gotten the dhcp option 242 to > work, and finally gotten the phone to the point where it asks for a username > and password. I defined these on the Asterisk server. I entered them on the

Re: [asterisk-users] Avaya 9608G and DHCP and TFTP and HTTP oh my

On Tue, Mar 06, 2018 at 05:36:04PM +, Thomas Peters wrote: > But please don't tell me the only way to program up each phone is via > the craft interface? > > Every other phone I've ever used requires a configuration file, which > has the MAC address of the phone as its name. The Avaya phones

Re: [asterisk-users] Half Off Topic Questions

On Tue, Mar 06, 2018 at 09:05:25AM +0100, Markus Weiler wrote: > we're just wondering, in German we call the different types of phone-numbers > (Geographic,mobile,national,VoIP...) > Rufnummerngassen (phone number alleys ;-) ) > Is there an english word for this? I'd call it something like

Re: [asterisk-users] Avaya 9608G and DHCP and TFTP and HTTP oh my

On Thu, Mar 01, 2018 at 02:46:31PM +, Thomas Peters wrote: > Right-- I've seen the Avaya document you cite below. It says "To > administer DHCP option 242, make a copy of an existing option 176" but > I don't have any example of option 176 or 242 to copy, and don't know > what to do to

Re: [asterisk-users] Avaya 9608G and DHCP and TFTP and HTTP oh my

On Wed, Feb 28, 2018 at 08:48:38PM +, Thomas Peters wrote: > I'd like to start configuring my Avaya 9608G phones for use on > Asterisk / FreePBX / PBX-In-a-Flash. I'm using a variety of other > phones on my system without major issues. > > I've read the discussion back in March, May and

Re: [asterisk-users] asterisk mysql contacts

On Wed, Jan 17, 2018 at 03:16:04PM +0200, Atux Atux wrote: [asterisk dialplan mysql] > I would like to ask if there is a way to implement this easily in my > dialplan, please. The answer is: yes If you'd search for "asterisk dialplan mysql", you'get something like

Re: [asterisk-users] remote Asterisk console

On Tue, Jan 16, 2018 at 06:19:30PM +0100, Paul Neuwirth wrote: > Thank you both. That was (most likely) what I was looking for - but > still some worries about sending plaintext passwords... The AMI interface can use a Challenge-Response mechanisme for logins, if you are this concerned you should

Re: [asterisk-users] Digium G100 and CID Dropping First Digit.

On Mon, Jan 15, 2018 at 10:41:27PM +, David Klaverstyn wrote: > port1 < Calling Number (len=12) [ Ext: 0 TON: National Number (2) NPI: > ISDN/Telephony Numbering Plan (E.164/E.163) (1) > port1 < Presentation: Presentation allowed of > network provided number (3)

Re: [asterisk-users] Can't install package asterisk-dbgsym on Stretch

On Fri, Dec 08, 2017 at 06:11:47PM +0100, Olivier wrote: > 1. Is this a bug in debian-debug repo ? If positive, should I file a bug > report ? > > 2. Is correct to understand that to get DONT_OPTIMZE, BETTER_BACKTRACE and > so on options compiled in, I must recompile anyway ? As far as I know

Re: [asterisk-users] Gerrit usage?

On Fri, Sep 29, 2017 at 12:27:53PM -0300, Joshua Colp wrote: > > "git checkout -b 13" appears to fix this. > > This did not create a branch from 13. This created a branch named "13" > from the branch you were on, which was most likely master. That is why > your "git review" is not working as you

[asterisk-users] Gerrit usage?

I'm trying to figure out how to commit some code for review. Following: https://wiki.asterisk.org/wiki/display/AST/Gerrit+Usage Created a ssh alias. Cloned using: "git clone ssh://asterisk/asterisk" Set name and email. Installed the gerrit commit hook: "git review -s" Try to change to asterisk 13

Re: [asterisk-users] ERROR during high volume MoH dialplan

On Thu, Aug 31, 2017 at 05:54:43PM +, Joseph Smith wrote: > > So I am looking for a better way to allow several thousand callers to listen > to this IVR menu at the same time. > An alternative that comes to mind is to have 1 conference with 1 channel playing MoH in it and then add callers

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] 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] 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] 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. >

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 > >

[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] 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] OT: Explain where mailing list bouncing comes from ?

On Fri, Jun 16, 2017 at 08:38:59AM +0100, J Montoya or A J Stiles wrote: > > Whatever has been done, if anything, isn't working effectively. At this > > point I'd like to see some response from the mailing list admin about any > > root-cause efforts, AFAIC this is starting to smear the

Re: [asterisk-users] Is this the future of telephony?

On Thu, Jun 15, 2017 at 08:56:29PM -0400, Christopher van de Sande wrote: > I just setup an anonymous endpoint in pjsip.conf and a context that > forwards to $EXTEN and when I setup the correct SRV records, it seems > that any SIP client that's smart enough can just dial my SIP/email > address.

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] CallerId presence issue

On Wed, Jun 14, 2017 at 10:18:19AM -0400, Mike wrote: > I have a PRI coming in PBX_A and PBX_A is connected to PBX_B via SIP. > PBX_A gets PRI calls on a 4 port Digium card, and each call naturally has > its own callerid values and presence. I pass on those calls to PBX_B via > SI, and I'm trying

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] OT: Explain where mailing list bouncing comes from ?

On Mon, Jun 12, 2017 at 09:07:31AM +0200, Olivier wrote: > Lately, I'm receiving emails asking me to re-enable my list subscription > due to "excessive bouncing". > > What does this exactly mean and why am I receiving this ? > Beside re-enabling my subscription, what can I do to improve things ?

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

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 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] 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] 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] Extensions of sip trunk

On Mon, Jun 05, 2017 at 06:10:50PM +0200, Hans-Peter Jansen wrote: > ; matches 12345678099, too > exten => _1234567800,1,Dial(SIP/int) > > Except from SIP invite with tcpdump: > > INVITE sip:12345678@provider:5060 SIP/2.0 > From: ;tag=as6bc7cbbc > To:

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

On Mon, Jun 05, 2017 at 01:08:17PM -0400, James B. Byrne wrote: > This is likely the issue surrounding mailing lists rewriting headers > and/or modifying messages bodies or simply re-transmitting messages as > the original sender from an unapproved domain. This was discussed at > length on the

[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] 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 >

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] 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] 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] 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 <->

[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] 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] 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] 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

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] 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] 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] 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] 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] 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] 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] [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] 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] 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] 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

[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] 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

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

[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] 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

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] 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] 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] 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

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