[asterisk-users] Choppy sound while converting alaw to ulaw

Hi there I europe alaw is usual. I have a SIP Phone which perferes ulaw. When my * box has to transcode alaw to ulaw the sound get's one way choppy. (alaw = ulaw is choppy, ulaw = alaw is fine). I managed to fix the issue by forcing my SIP phone to use alaw only, but is this a know issue with

[asterisk-users] Zapateller not playing audio via SIP Trunk?

,n,Hangup() Early Audio Announcement is played, but then again as soon as Zapateller is executed, it 'hangs'. Any idea what causes Zapateller to hang if it should play early audio via a SIP Trunk? Mit freundlichen Grüssen Benoit Panizzon -- I m p r o W a r e A G-System Services

Re: [asterisk-users] Zapateller not playing audio via SIP Trunk?

exten = i,1,Zapateller() Same happens if I use PlayTones(info) instead of ZapaTeller(). Same happens if I use Progress() before ZapaTeller or Playtones. Mit freundlichen Grüssen Benoit Panizzon -- I m p r o W a r e A G-System Services

[asterisk-users] Scratchy Audio with Asterisk 1.2.4 over IAX on FreeBSD?

shows that problem, gsm doesn't work at all, but that could be a codec problem of the 1.2.4 gsm implementation) Playing around with the IAX jitterbuffer settings does not affect the scratching sound in any way. Any idea what the cause could be? Mit freundlichen Grüssen Benoit Panizzon -- I m p

[asterisk-users] SIP Redirect

more transparently by a redirect. Mit freundlichen Grüssen Benoit Panizzon -- I m p r o W a r e A G-System Services __ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 PrattelnFax +41 61 826 93 01 Schweiz

[asterisk-users] Voicemail, how to localize date in email notifications?

:%S produces English Day and Month Names within our email sent in german. Can this be changed without altering the System Locale? Mit freundlichen Grüssen Benoit Panizzon -- I m p r o W a r e A G-System Services __ Zurlindenstrasse

[asterisk-users] BRI: Asterisk disconnecting on 'call diverted' message?

812 9b2 ] 2 Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) 2 Ext: 1 Cause: Unknown (27), class = Normal Event (1) ] -- Hungup 'Zap/4-1' Mit freundlichen Grüssen Benoit Panizzon -- I m p r o W a r e

[asterisk-users] Multiple 'routes' to extension in different contextes. How to influence search oder?

to change it everywhere. Regards Benoit Panizzon -- I m p r o W a r e A G-System Services __ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 PrattelnFax +41 61 826 93 01 Schweiz Web http

[Asterisk-Users] Server freeze with meetme and sip GSM users and ztdummy

connected to meetme. And known issues? Benoit Panizzon -- I m p r o W a r e A G-System Services __ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 PrattelnFax +41 61 826 93 01 Schweiz

[Asterisk-Users] Set CallerID to a specific Queue Member

/provider-2/zz Now I must set the correct callerid thich is registered at each provider for outgoing calls and those are different callerids for provider-1 and provider-2. How can I solve this Problem? Mit freundlichen Grüssen Benoit Panizzon -- I m p r o W a r e A G-System Services

[Asterisk-Users] Re: Server freeze with meetme and sip GSM users

Hi Brent Anyone ever seen MeetMe cause * to crash? Specifically, it happens consistantly if someone begins to enter a conference and then decides to hangup while Allison is introducing them - like playing back conf-onlyperson. This has been seen with the MeetMe participant connecting via IAX

[Asterisk-Users] Anonymous sip calls getting into wrong context?

Hi all Maybe somebody has an idea. I'm tracing a very strange phenomena... I've a connection from Asterisk to a SIP PBX. Most calls have a caller ID. Some International calls don't have any. Now it looks like those calls without caller ID never get to the context where incomming calls from

Re: [Asterisk-Users] Re: Server freeze with meetme and sip GSM users

with english language files... Benoit Panizzon -- I m p r o W a r e A G-System Services __ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 PrattelnFax +41 61 826 93 01 Schweiz Web http

[Asterisk-Users] * Meetme Freeze patch found

Hi all Apparently there is a patch for those 1.2.4/5 MeetMe Freezes: http://bugs.digium.com/view.php?id=5884 Haven't tried it out yet. Benoit Panizzon -- I m p r o W a r e A G-System Services __ Zurlindenstrasse 29 Tel

Re: [Asterisk-Users] * Meetme Freeze patch found

On Friday 24 March 2006 16:05, Benoit Panizzon wrote: Hi all Apparently there is a patch for those 1.2.4/5 MeetMe Freezes: http://bugs.digium.com/view.php?id=5884 Haven't tried it out yet. I can now confirm: No freezes/crashes anymore since I applied the patch. -Benoit

[Asterisk-Users] How to send announcement after called has picked up the phone?

Hi I would like to send a text to the called person when he picks up the phone before the call gets connected through. Is there a way to do this? Example: I'm registered to multiple SIP providers. They come in to a context each and then get through to my phone. Now I would like to send myself

[Asterisk-Users] zaphfc on an 'actual' asterisk?

Hi all I don't manage to get asterisk 1.2.5 or 1.2.6 running with the zaphfc driver The scripts from junghanns.net do download a very old libpri and asterisk version which is too buggy for me to use. Isn't there an acutal patch to get zaphfc support in *? -Benoit-

Re: [Asterisk-Users] zaphfc on an 'actual' asterisk?

Hello 1; you stay with the current bristuff (a somewhat older zaptel+asterisk, but is this really making a difference?) Oh yes. Up to 1.2.4 there is a bug in the way * replies to as SIP Options request which makes it impossible to connect it to some commercial PBX. Up to 1.2.6 there is a bug

Re: [Asterisk-Users] mISDN Problem

6240 1 isdn mISDN_l2 36064 0 w6692pci 23692 0 mISDN_core 74944 2 mISDN_l2,w6692pci Mit freundlichen Grüssen Benoit Panizzon -- I m p r o W a r e A G-System Services

[Asterisk-Users] misdn timeout?

Hi all I have a very strange problem here... I use a hfc-s card with mISDN in NT mode with an ISDN Phone connected. When I make a call, the phone rings two or three times and then misdn runs into a timeout... I don't know where to set that timeout, but it's way to short for the called to pick

Re: [Asterisk-Users] misdn timeout?

P[ 1] I IND :TIMEOUT oad:0010618115711 dad:4680041618269314 # Sudden Timeout?!? Uhm, I just found the problem myself after a bit more testing. Apparently my phone has a timeout of about 10 seconds in which it waits for any reply. Well you head that it's ringing, but it's not signaled as such

Re: [Asterisk-Users] bristuff for * 1.2.6/zaptel 1.2.5

) that a new PRE bristuff was about to come out, supporting 1.2.6. For now I have switched to mISDN. Mit freundlichen Grüssen Benoit Panizzon -- I m p r o W a r e A G-System Services __ Zurlindenstrasse 29 Tel +41 61 826 93

Re: [Asterisk-Users] bristuff for * 1.2.6/zaptel 1.2.5

miss functions like setting the time of the phones, call forwarding etc... but I suppose not all of these would also work on the ZapHFC drivers... Mit freundlichen Grüssen Benoit Panizzon -- I m p r o W a r e A G-System Services

[Asterisk-Users] zaphfc NT Mode. Extension not recognized...

Hi all I finaly set up a second * with two ZapHFC Cards. One in TE the other in NT mode. So I have a 1.2.5 Asterisk to run Meetme etc... and a 1.2.4 Asterisk to run all that Zaptel stuff. First I used mISDN on 1.2.5 which worked, but sometimes had strange behaviour. So my hope was that zaptel

[Asterisk-Users] Got SIP response 302 Moved temporarely

provider tryes to redirect to. Any known issues? Mit freundlichen Grüssen Benoit Panizzon -- I m p r o W a r e A G-System Services __ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 PrattelnFax +41 61 826

[Asterisk-Users] extensions.conf - switch = statement?

between these machines' as 'switch' is explained in the examples. But how do I do that exactly? Mit freundlichen Grüssen Benoit Panizzon -- I m p r o W a r e A G-System Services __ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH

Re: [Asterisk-Users] zaphfc NT Mode. Extension not recognized...

Hi all I managed to figure out where the problem is... zapata.conf channel = has to bee the last statement per channel definition. So if you specify overlapdial=yes after channel = this has no effect. You need to stop and restart asterisk after changes to zapata.conf. Reload does not seam to

[Asterisk-Users] Dial Plan Problem with extensions ringing multiple phones connected on different * servers

Grüssen Benoit Panizzon -- I m p r o W a r e A G-System Services __ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 PrattelnFax +41 61 826 93 01 Schweiz Web http://www.imp.ch

Re: [Asterisk-Users] How to set busy

On Sunday 09 April 2006 06:02, Miles Scruggs wrote: For multiline phones how do you set SIP channels to busy. For instance if SIP/101 is on a call then dial would return busy. Right now it just starts ringing on line X, and stacks up from there. I suppose incominglinit=1 in the sip.conf of

[Asterisk-Users] User Defined VoiceMail announcement?

PBXes) that the VoiceMail user could record his own announcement? (like, hello, this is the Voicebox of John Smith, please leave a message after the tone). Mit freundlichen Grüssen Benoit Panizzon -- I m p r o W a r e A G-System Services

[Asterisk-Users] Dial 'R' option gone?

Hi After migrating from 1.2.4 to 1.2.5 I noticed that: show application dial does not show the 'R' option anymore. Has this become an undocumented feature or has it gone completely? Mit freundlichen Grüssen Benoit Panizzon -- I m p r o W a r e A G-System Services

Re: [Asterisk-Users] Dial 'R' option gone?

On Friday 28 April 2006 15:32, Eric ManxPower Wieling wrote: What does the R option do? Indicate 'Ringing' as soon as the called party indicates 'Ringing'. The 'r' option indicates 'Ringing' as soon as the connection is built, even if the called party is not yet ringing. With some SIP

[Asterisk-Users] Dropped SIP connections never being closed?

Hi all I have noticed an interresting problem with Wireless SIP Connections. When a Phone gets out of reach during a call (for example into a MeetMe Conference) of course the connection gets lost. The Phone hangs up. But 'show channels' still shows that call, and the user is still in the

[Asterisk-Users] How to set overlap dial timeout in bristuff zaptel?

Hi all There seam to be a very short timeout waiting for digits being dialed. (about 6 seconds). Is there a way to increase that time? I have a phone with integrated address book and my fingers are just not fast enough to open the menue, select an entry and hit 'dial'. -Benoit-

[asterisk-users] Asterisk Voicemail Realtime and 'VirtualBoxing'

the email settings per voicemail context together with a realtime vm config? Mit freundlichen Grüssen Benoit Panizzon -- I m p r o W a r e A G- __ Zurlindenstrasse 29 Tel +41 61 826 93 07 CH-4133 PrattelnFax

[asterisk-users] VoiceMail customizing

how messages are played via voicemail.conf? Mit freundlichen Grüssen Benoit Panizzon -- I m p r o W a r e A G- __ Zurlindenstrasse 29 Tel +41 61 826 93 07 CH-4133 PrattelnFax +41 61 826 93 02 Schweiz

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

there is an unavailable message. Where do I have to poke at the source? Kind regards Benoit Panizzon -- I m p r o W a r e A G- __ Zurlindenstrasse 29 Tel +41 61 826 93 07 CH-4133 PrattelnFax +41 61 826 93 02

[asterisk-users] Early audio SIP sequence order question

Session Progress instead of 180 Ringing? Kind regards Benoit Panizzon -- I m p r o W a r e A G- __ Zurlindenstrasse 29 Tel +41 61 826 93 07 CH-4133 PrattelnFax +41 61 826 93 02 Schweiz

Re: [asterisk-users] Early audio SIP sequence order question

of it didn't yet receive a 183 or 200 message, or is the carrier doing wrong in sending early audio without 183? Kind regards Benoit Panizzon -- I m p r o W a r e A G- __ Zurlindenstrasse 29 Tel +41 61 826 93 07 CH-4133

Re: [asterisk-users] Early audio SIP sequence order question

more than one endpoint and more than one is sending early audio, which one do you forward? I think nobody tought about that issue. Well as long as one is being forwarded that would be ok for our case :-) Kind regards Benoit Panizzon -- I m p r o W a r e A G

[asterisk-users] SIP Diversion RDNIS - how to get reason parameter?

Hi out there To play the correct announcement in app_voicemail I whould be able to read the SIP Diversion Reason which ist sent by another PBX: Invite contains: Diversion: sip:+41315995003@157.161.10.190;reason=no- answer;privacy=off;counter=1 Asterisk Logs: RDNIS for this call is is

Re: [asterisk-users] SIP Diversion RDNIS - how to get reason parameter?

in a temporary variable __SIPDIVERSIONREASON but not in a variable useable in the dialplan. Kind regards Benoit Panizzon -- I m p r o W a r e A G- __ Zurlindenstrasse 29 Tel +41 61 826 93 07 CH-4133 Pratteln

[asterisk-users] cseq decreasing = 500 Server Error

Benoit Panizzon -- I m p r o W a r e A G- __ Zurlindenstrasse 29 Tel +41 61 826 93 07 CH-4133 PrattelnFax +41 61 826 93 02 Schweiz Web http://www.imp.ch

[asterisk-users] Prevent Asterisk from setting CALLERID(name) or unsetting CALLERID(name)

a CALLERID(name) if there is none? Id did try to set ${CALLERID(name)=} but that resulted in From: sip... and the displaying of this empty string on the subscribers phone. Is there a way to completely remove the CALLERID(name) like (UNSET({CALLERID(name))? Kind regards Benoit Panizzon -- I m p r o

[asterisk-users] chan_sip.c:3641 retrans_pkt: Retransmission timeout

the Asterisk ignores them. I did try to get more information with debug and verbose level set to '99', but I don't see more messages Does anyone have a clue, why acks could be not accepted by asterisk 1.8.10.0 ? The other way round (asterisk = c3) the calls work fine. Regards Benoit Panizzon

[asterisk-users] Invite + decreasing sequence number = 500 Error?

who knows? Benoit Panizzon -- I m p r o W a r e A G- __ Zurlindenstrasse 29 Tel +41 61 826 93 07 CH-4133 PrattelnFax +41 61 826 93 02 Schweiz Web http://www.imp.ch

Re: [asterisk-users] Invite + decreasing sequence number = 500 Error?

them the _MUST_ part of section 22.2 Thanks Benoit Panizzon -- I m p r o W a r e A G- __ Zurlindenstrasse 29 Tel +41 61 826 93 07 CH-4133 PrattelnFax +41 61 826 93 02 Schweiz Web

Re: [asterisk-users] Invite + decreasing sequence number = 500 Error?

) SIP/2.0 500 Server error Well as I see it, the C3 PBX just generates plain random CSeq Numbers. Regards Benoit Panizzon -- I m p r o W a r e A G- __ Zurlindenstrasse 29 Tel +41 61 826 93 07 CH-4133 Pratteln

[asterisk-users] How to log caller IP address in the CDR?

it is too late now. But is there a way to get the IP Address of the SIP Client being logged in each CDR? Kind regards Benoit Panizzon -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join

Re: [asterisk-users] How to log caller IP address in the CDR?

set: alwaysauthreject=yes And got a script to scan the logfile all 15min to firewall IP addresses which excessively try to login. You're always smarter after the incident :-/ Benoit Panizzon -- _ -- Bandwidth and Colocation

[asterisk-users] t.38 fax over IAX2?

Hello Let's assume we have this situation: Call => SIP TSP => Asterisk1 => IAX2 => Asterisk2 => SIP/ATA => Fax I have two Asterisk Servers in two branch offices, which are interconnected by IAX2 and the Switch functionality. Asterisk1 is connected to the public phone network via a SIP provider

[asterisk-users] SIP Trunk over Proxy (matching ip of outbound proxy in incomming calls)

Hello List I work at an SIP Provider and we have added and SBC in front of our Voice Switch to protect it. This requires all our SIP Trunk customers to register via a 'proxy'. I struggle with Asterisk to work over a proxy. This is what I have done so far. register =>

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

Hi all I know, a fairly old asterisk installation. Is there any way to debug T.38 handshaking issues? We have a C3 Voice Switch with link to the Asterisk server. I see this Dialogue: C3 => Asterisk => Invite g711 <= 200OK C3 detects Fax and send re-invite => Invite T.38 Version:0

[asterisk-users] CallerID(num-pres) not set during incomming call

Hello List I can set CALLERID(num-pres)=prohib on a sip channel and asterisk is setting the headers more or less correctly (PAI Header is missing maching the call untrackable, which is a bit odd). But when asterisk is handling an incomming call from: From: Anonymous

[asterisk-users] Now to set contact username and from username idependently

Hello Finally I figured out, how our SBC does matches invites to registrations with the Contact header. But now I run into a Problem: How do I set the contact header of an invite different to the From header? INVITE sip:called-id@URI SIP/2.0 Via: SIP/2.0/UDP IP:5060;branch=z9hG4bK495c70cc

[asterisk-users] IAX2 via IPv6, no packets being sent!

Hello List I have two IAX2 peers reachable with IPv6. They consider them self unreachable. If I do a 'iax2 set debug on', I see asterisk pretending to send POKE packets to the IPv6 address of the peer. If I sniff on the interface, I don't see those packets. Is there a known issue? Version

[asterisk-users] Queue, no announcement being played at all

Hi List I have a very strange problem. I was using queues a while ago with an asterisk 1.2 or so and announcements were working fine more or less out of the box. Now I am once again trying to set up a queue with Version 13.14.1 an not matter what I do, I don't get the announcements to be played.

[asterisk-users] Outbound Calls via Proxy to use Call ID from registration

Hello List > I work at an SIP Provider and we have added and SBC in front of our > Voice Switch to protect it. Well using two peers for incomming and outgoing calls solve the previous issue. Now I have a new one. The SBC in use needs to match incomming calls from the asterisk with the call id

[asterisk-users] pjsip multiple transports for one endpoint (dual stack) ipv6

Hi List I have stumbled over the next question google didn't answer. I have a dual-stack environment, ipv6 and ipv4. With chan_sip asterisk was listening on ipv6 and ipv4 simultaneously. I did try to define to have pjsip listen to the ipv6 address including ipv6 mapped ipv4 addresses:

[asterisk-users] pjsip Transfer 'Failed to parse destination uri'

Hi Richard > That could be possible and would be a bug in chan_sip. Ok, so I switched to PJSIP to see if this behaves differently So ip do a Transfer(PJSIP/${DESTNUMBER}@trunk) And this results in: Failed to parse destination URI '[destnumber scrubber]' for channel PJSIP/trunk-0011 Do I

Re: [asterisk-users] pjsip Transfer 'Failed to parse destination uri'

Ok, answering myself: Asterisk 13.14.1~dfsg-2+deb9u2 Apparently suffers the pjsip transfer bug described @ https://reviewboard.asterisk.org/r/4316/diff/ Specifying the full URI: Transfer(PJSIP/sip:${DESTEXTEN}@trunk) does resolve the URI parsing problem and is sending back the 302 message

[asterisk-users] SOLVED! Re: pjsip subscribe (presence) always returns: No matching endpoint found

Hi List Just in case someone else runs into the same problem migrating from chan_sip to res_pjsip. In chan sip you did define the voicemail variables in the peer section. I did configure most of that stuff into the endpoint of pjsip, including: mailboxes= voicemail_extension= Well, after

Re: [asterisk-users] pjsip subscribe (presence) always returns: No matching endpoint found

Hi Joshua > The chan_pjsip module doesn't prevent that. You'd need to provide the > full SUBSCRIBE now that it is actually finding the endpoint and coming > in. Ok, let's see if we can solve the mystery.. pjsip.conf [endpt-home](!) type=endpoint disallow=all allow=g722 allow=alaw allow=gsm

Re: [asterisk-users] How to correctly set REDIRECTING to indicate diversion reason

Hi Richard Thank you > You need to set more redirecting information [1]. > > In sip.conf send_diversion=yes needs to be in effect. You also need > to setup > the from party id information (at least the from number) to indicate > where you > are redirecting from. You should also increment the

[asterisk-users] pjsip subscribe (presence) always returns: No matching endpoint found

Hello List I am in the progress of migrating from chan_sip to pjsip. I fear I have missed something on how hints need to be specified for pjsip. For chan_sip I have configured sip.conf subscribecontext = localuser and in the dialplan I set: [localuser] exten => 11,hint,SIP/11 Now if a

Re: [asterisk-users] pjsip subscribe (presence) always returns: No matching endpoint found

Hi Joshua thank you for the quick reply > Have you checked the Asterisk console when PJSIP is loaded to see if > the endpoint did not load for some reason? Does it show up in "pjsip > show endpoints"? Yes, the endpoint shows up. Endpoint: 11/(scrubbed from mail)

[asterisk-users] Ringing (180) no SDP to progress(183) with SDP transition => no audio.

Dear List I am testing various early audio scenarios with different voice IC's, phones and pbxes. In Switzerland, when you operate a value added number, you have to announce the price of the call, usually in early audio, before the call is established. In 'dialplan' terms this would be: exten

[asterisk-users] How to correctly set REDIRECTING to indicate diversion reason

Hello List Next question where google did not spit out an unsable answer. When redirecting a call with Transfer, I would like to correctly indicate the reason. I did try this: exten => XX,1,NoOp(Call to ${EXTEN} from ${CALLERID(all)}) exten => XX,n,Dial(SIP/ZZ) exten =>

Re: [asterisk-users] pjsip rtp_ipv6=yes but endpoint registered via ipv4 (IP4 contact infor)

Hi Jushua > The rtp_ipv6 option is not needed, in current versions things will > automatically be updated to reflect the signaling. Remove it and give > it a try. The option itself actually had the bug that you are seeing. Ok, commented out rtp_ipv6 in the config and did try again: IPv6

Re: [asterisk-users] PJSIP: identify endpoint by authentication username?

Hi George > [global] > endpoint_identifier_order = auth_username,username,ip,anonymous > > [endpoint_x] > identify_by = auth_username Thank you, I missed that config option, works perfectly! Mit freundlichen Grüssen -Benoît Panizzon- -- I m p r o W a r e A G-Leiter Commerce Kunden

[asterisk-users] pjsip rtp_ipv6=yes but endpoint registered via ipv4 (IP4 contact infor)

Dear List I fear I stumbled over a bug in asterisk 13.14.1. My 'phones' are roaming around, sometimes some are connecting from ipv6 enabled networks, another time they are not. If a connection is ipv6 I would prefer to use ipv6 to avoid ipv4-nat problems. I have not specified a transport in

[asterisk-users] PJSIP: identify endpoint by authentication username?

Dear fellow list readers This is the situation: ISDN Devices => Patton ISDN to SIP GW => Asterisk PJSIP The Patton GW resides on a dynamic IP address, so I cannot really use match=ip in the identify section. The Patton does not send a line parameter. The ISDN Devices behind the patton have

[asterisk-users] Weird 'hairpin' call rtp audio problem

Hello List Asterisk 13.14.1 in use with pjsip stack. On the remote side is a SBC which performs some 'nat' detection. I suppose this means the SBC listens from where it is getting RTP data and then replies to that ip. As long as the asterisk is initiating the call this is fine, the asterisk

Re: [asterisk-users] Weird 'hairpin' call rtp audio problem

Hi Joshua > The "rtp_keepalive" option can be used to have the RTP stack send an > RTP packet out. Try that and see what happens. Once again 'bullseye' that fixed the problem. Thank you! Mit freundlichen Grüssen -Benoît Panizzon- -- I m p r o W a r e A G-Leiter Commerce Kunden

[asterisk-users] What is the status of world wide e164 DUNDI

Hello List I have a still two connected DUNDI peers, but they seem to flap from time to time. A couple of years ago I was able to look up quite some, mostly free call numbers via DUNDI all over the world and I als saw incomming lookups. But not anymore. I wonder if I am stranded on a no longer

[asterisk-users] To Header instead of Request URI based routing

Dear List It looks like the common way to to sip signaling over a trunk is: In the Request URI, return the 'Register' Contact. In the To: Header, send the destination number. Unfortunately, asterisk with pjsip (i did not try chan_sip) does expect the dialed extension as request uri and does

[asterisk-users] Asterisk receiving 415 Unsupported Media Type upon T.38 invite behaving absolutely weird.

Hello I am hunting Fax Problems. Now I have come across a situation on which, I fear asterisk behaves in a wrong manner.. A T.38 enabled ATA is connected to the asterisk and receiving a call from a non T.38 capable endpoint. The ATA is detecting the CED tone and initiating a T.38 re-invite.

Re: [asterisk-users] Weird 'hairpin' call rtp audio problem

Hi Joshua > > The "rtp_keepalive" option can be used to have the RTP stack send an > > RTP packet out. Try that and see what happens. > > Once again 'bullseye' that fixed the problem. Thank you! Now a customer with and FreePBX 2.9.0 (Asterisk 1.8.20.1) ran into the same issue with our SBC. I

Re: [asterisk-users] How to enable TLS debugging or verbose logging with pjsip

Well, when testing with: $ openssl s_client -connect tls-host:5061 I get a successfull TLS handshake and connection. So I suppose asterisk is configured correctly with TLS. I did re-check the cipher list and also this seems to match on the SPA112 and Asterisk. So I am puzzled why the SPA112

Re: [asterisk-users] Blacklist failed attempts

Hi You could do somethink like this in Perl: #!/usr/bin/perl -w use strict; use warnings; my (@failhost); my %currblocked; my %addblocked; my $action; open (MYINPUTFILE, "/var/log/asterisk/messages") or die "\n", $!, "Does log file file exist\?\n\n"; while () { my ($line) = $_;

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

Hey List I sometimes use our asterisk server to do some debugging or other PBX and SBC. Now we have a case where a PBX is replying an incomming invite with 180 ringing immediately. It looks like the SBC does not accept this. According to my understanding of the RFC 3261 any provisional (aka

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

Hi Tryba > A (very) dirty workaround would be to drop these packets with iptables > (assuming Linux as OS), something like: > > iptables -t raw -I OUTPUT -p udp -d ipaddrofpbx -m string --algo bm > --from 0 --to 32 --string "SIP/2.0 100 " -j DROP > > Don't try it with TCP :) :-) Indeed, this

[asterisk-users] How to enable TLS debugging or verbose logging with pjsip

Dear List I try to get my clients to connect via TLS. First I did try Snom M9 phones. After looking at the Wireshark TLSv1 Handhake it became obvious, that the M9 only supports old RC4 and similar ciphers, that are not supported by openssl anymore. So now I get my hands on a Cisco SPA112 ATA,

[asterisk-users] RFC about SIP 'To' header after call diversion?

Hi List I'm struggling to find the correct RFC which "exactly" defines how a SIP Invite has to look like after a call has been diverted. Especially what the content of the To: header field has to be. Example call flow: Alice calls Bob who diverts to Carol. Alice => Bob Invite:

[asterisk-users] pjsip vs chan_sip: Where is callerid(num) taken from?

Dear List We are renewing our voicemail server and by this occasion I am migrating from chan_sip to pjsip. I have come to a problem I have not experienced on other pjsip examples. Switzerland was heavily SS7 based in the past. So usually you have a Network provided A Number, which is mapped to

Re: [asterisk-users] pjsip vs chan_sip: Where is callerid(num) taken from?

Ok, just figured it out, looks like pjsip uses some reversed trust logic... PAI contains the network provided screened number, the one which can be trusted and used for billing purposes and similar. From contains the generic number, which should be displayed, but which is user provided and

[asterisk-users] Dial(${PJSIP_DIAL_CONTACTS(Alice)} & ${PJSIP_DIAL_CONTACTS(Bob)}) how not to fail if one endpoint has no registered AOR?

Dear List It's probably been more than a year now I switched from chan_sip to pjsip. pjsip works much cleaner than chan_sip. But! I have come across a Problem I was not able to solve with Asterisk Dialplan Logic. With pjsip an endpoint can have multiple AOR, so you need to expand them with

Re: [asterisk-users] Dial(${PJSIP_DIAL_CONTACTS(Alice)} & ${PJSIP_DIAL_CONTACTS(Bob)}) how not to fail if one endpoint has no registered AOR?

> What about to put eveything in a variable and the remove the last > character if it equal & Yes, I considered this... What if you dial three endpoints and the middle one (or last one) is empty? You would also need to remove the first & and any double & within that string. Is it faisable with

[asterisk-users] Two interfaces, pjsip, 180 ringing contains wrong contact IP

Hi Gang I have stumbled over a strange issue with Asterisk 13.18.3 I have two interfaces, two different IP Addresses. One facing to the internet, and one facing to am internal voice lan. Therefore I defined two different transports and endpoints: [transport-udp-internal] type=transport

[asterisk-users] On Register, run a script, validate source IP

Hi Gang To increase security against phished passwords and similar attacks, we consider offering customers to define IP ranges (or GeoIP locations) from which their dynamic registrations are being accepted. I can already look at the source IP in the dial plan, so no issue with validate an INVITE

Re: [asterisk-users] pre-dial handler, how to access variables from calling channel?

Hi Tony > See https://wiki.asterisk.org/wiki/display/AST/Variable+Inheritance Thank you, exactly what I was looking for! Mit freundlichen Grüssen -Benoît Panizzon- -- I m p r o W a r e A G-Leiter Commerce Kunden __

Re: [asterisk-users] On Register, run a script, validate source IP

Hi Sebastian > That would require your script to update sip.conf dynamically and reload the > config for each time user wants to update their accepted location. Hmm, maybe using asterisk realtime and attempting to put the config into a database would be worth an approach. Until now we only use

[asterisk-users] Check other calls on same endpoint (validate / screen customer supplied Diversion / From header)

Hi Gang Yes, big project on the rise to do things better / more flexible than our existing commercial TSP switch. During call screening process, we would like to allow customers to send the original callingID in a attended call diversion scenario. From the Voice Switch point of view, there are

[asterisk-users] pre-dial handler, how to access variables from calling channel?

Hi List Implementing screening and routing I have stumbled over this issue: [pbx-router] exten => s,1,NoOp(ROUTER FROM: ${CALLERID(Number)} TO: ${DESTINATION}) same => n,Set(SOURCE=${CHANNEL(name)}) same => n,Set(PAI=${PJSIP_HEADER(read,P-Asserted-Identity)}) same =>

Re: [asterisk-users] On Register, run a script, validate source IP

Hi Jöran > for me it sounds like you need an SBC. > We use Kamailio in order to check users IP Addresses. There are modules > like "permissions" in kamailio what could do this. As well there are pike > checks, sanity checks and a bunch of other useful tools. You are absolutely right. We are on a

Re: [asterisk-users] Global number rewriting rules affecting ALL headers?

Hi Joshua I had a shot at your suggestion, bug still no success. I fear the 181 is sent before the macro is called. I want to change the Diversion Header in the 181 message sent back to the caller to put the number it contains in the correct e164 format (stripping the 0 and adding +41 for

[asterisk-users] Global number rewriting rules affecting ALL headers?

Hi List One more Problem I stumbled upon. Using Asterisk in a TSP environement. Incomming IC Calls are e164 and have a NPRN (Routing Number) prefixed. Example: +419805561599 +41 country prefix 98055 Routing Prefix 61599 effective phone number Calls routed to Customers need to be put

[asterisk-users] trust_id_inbound=yes but take CallerID(Num) from From: not from PAI

Hi Gang Next Problem which occurs. In Switzerland this is the common using form SIP Signaling: P-Asserted-Identity: Contains the provider provided and screened phone number which is the 'legal' origin of the call. The origin which is to be billed for the call. If the caller has a DDI Range,

Re: [asterisk-users] Global number rewriting rules affecting ALL headers?

Quick update. I guessed right. I had put the call to the subrouting on the 'local' channel which is created after the call is being redirecting. If i put it on the calling channel and setting RDNIS to the correct value, the corrected phone nuber is transmitted to the calling party via Diversion

Re: [asterisk-users] bug in pjsip trust_id_outpound?

Hi Gang If anyone else stumbles over the same Problem. This is how I solved it for now: On the IC Trunk: trust_id_inbound=no => Makes sure the CallerID is taken from the From Header. trust_id_outbound=yes => Does nothing useful, maybe a bug? send_pai=no On the incoming call, you have to pull

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