Re: [asterisk-users] Asterisk 13 / chan_sip / registration after reject

Hello > > How do I achieve the same with chan_sip? > We run a cron script each 10min who will check the registration state > and send a register if not registered. Well it's a simple CPE which needs to be autoprovisioned via either a tftp config file or TR69. So that cronjob somehow would

[asterisk-users] Asterisk 13 / chan_sip / registration after reject

Hi List We have some CPE which run an embedded asterisk 13 with chan_sip. Unfortunately, when a registration is rejected, those stop trying. I am familiar with pjsip which allows to configure: auth_rejection_permanent=no How do I achieve the same with chan_sip? Mit freundlichen Grüssen

[asterisk-users] DUNDI anyone?

Hi Well it is well some time that my last DUNDI peer has become unreachable. I guess too many issues with spoofed numbers etc. But I am wondering, do people, especially larger entities like telcos, still use DUNDI? I know that in some Hamradio communities, DUNDI is used to interconnect PBXes,

[asterisk-users] Ping

My last post did not make it back or to the archive... testing... Mit freundlichen Grüssen -Benoît Panizzon- -- I m p r o W a r e A G-Leiter Commerce Kunden __ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln

[asterisk-users] Asterisk PJSIP setting don't fragment bit on UDP

Hi Gang I noticed, that when I enable multiple codecs and rtp encrypting (generating a large SDP) invites with credentials do not get through anymore. So sniffed the connection and found that the IP packets have the don't fragment bit set, causing a VDSL router with 1472 MTU in the path to

Re: [asterisk-users] Asterisk 16.16.1 crash upon receiving image 0 udptl t38 sdp

Hi Josh > This was a security issue[1] which was solved. > > [1] https://downloads.asterisk.org/pub/security/AST-2021-006.html Thanks, filing Bugreport with Debian, hopefully they will push 16.16.2 to security updates. Mit freundlichen Grüssen -Benoît Panizzon- -- I m p r o W a r e A G

[asterisk-users] Asterisk 16.16.1 crash upon receiving image 0 udptl t38 sdp

Hi List I can reproduce Asterisk 16.16.1 segfaulting in this situation: Asterisk configured with Application "ReceiveFax". Incoming call with SDP: v=0 o=prt-cbl-sbc1 1418830458 1418830459 IN IP4 157.161.X.X s=sip call c=IN IP4 157.161.X.X t=0 0 m=audio 11828 RTP/AVP 9 8 101 a=rtpmap:9

Re: [asterisk-users] Asterix 16, PJSIP outbound registration, SRV or NAPTR lookup

Just stubled over another example which resolved my question. > server_uri=sip:reg.example.com:5060 > client_uri=sip:testcont...@reg.example.com:5060 If you don't specify the port, asterisk DOES an SRV lookup. Mit freundlichen Grüssen -Benoît Panizzon- -- I m p r o W a r e A G-

[asterisk-users] Asterix 16, PJSIP outbound registration, SRV or NAPTR lookup

Hi Team I'm working on a scenario, where the registrar offers multiple instances that can handle registration: _sip._udp.reg.example.com has SRV record 0 0 5060 reg01.example.com _sip._udp.reg.example.com has SRV record 0 0 5060 reg02.example.com It looks like specifying:

[asterisk-users] Voicemail: don't play vm-intro if custom intro is recorded.

Hi Gang We migrated our voicemail system from asterisk 13 to 16 a couple of months ago. Right after the migration, we got the complaint that vm-intro is being played when the customer had recorded a own announcement. So I assumed we had replaced that file by a zero lenght one on the previous

[asterisk-users] How to correctly fork a CDR for billing in a call forwarding scenario?

Hi Gang I have not yet managed to find a solution to correctly generate CDRs for this situation: Alice calls Bob. Bob has call forwarding delayed 20s to Charlie. Charlie picks up immediately. exten => bob,1,DBget(cfwdly=CFDLY/${exten}); $cfwdly contains charlie same =>

[asterisk-users] Get PJSIP Endpoint Information via REST or similar API?

Hi Gang To get our customers more information on how they registered I am looking for a elegant way to get an information like the CLI command: pjsip show endpoint [endpoint] I had a got on ARI, but that basically only returns the information if an endpoint is online or not. Is there a API to

Re: [asterisk-users] Perl AGI: read variable with quotes

Hi Gang Thank you for the replies. I sorted this out. I got tricked by $AGI->verbose(Pai: $pai) which cripples the output. The variable passed on is complete. My regex to extract the phone number from that variable was broken when there was a quoted string before the URI. Mit freundlichen

[asterisk-users] Perl AGI: read variable with quotes

Hi Gang I have stumbled of this problem. I need the P-Asserted-Identity header in an AGI scrip. In the Dial-Plan I do: same => n,Set(PAI=${PJSIP_HEADER(read,P-Asserted-Identity)}) In the AGI I do: my $pai = $AGI->get_variable(PAI); This works fine, unless the PAI contains quotes:

[asterisk-users] PJSIP do not challenge 'options' without username. - silence 'notice' on console.

Hi Gang Mitel PBX use 'options' without username to monitor the connection. Therefore Asterisk PJSIP cannot match an unsername against an endpoint and prints a notice on the console. Is there a way to silence this kind of notice? I wonder if identify_by 'header' could solve the issue to match

Re: [asterisk-users] res_pjsip.c:3461 ast_sip_set_tpselector_from_transport_name: Unable to retrieve PJSIP transport 'transport-name'

Ok, answering myself... It looks as if registered endpoints are cached in a way which survives a full restart of asterisk. So after deleting the transport, there was still a cached registered endpoint present via that transport. As soon as the Registration expired, the error also disappeared.

[asterisk-users] res_pjsip.c:3461 ast_sip_set_tpselector_from_transport_name: Unable to retrieve PJSIP transport 'transport-name'

Hi Gang I gave up on running asterisk with two interfaces without it mixing up the ip addresses. So I have removed one transport definition from pjsip.conf Now * keeps complaining: res_pjsip.c:3461 ast_sip_set_tpselector_from_transport_name: Unable to retrieve PJSIP transport 'transport-name'

Re: [asterisk-users] Solved: Re: Asterisk 13.18.3 PJSIP. Wrong Port in Contact Header in Reply to REGISTER?

Well, not so solved unfortunately... Now I am back to where I have the situation the Asterisk sends out 183 Media Progress from one interface, containing a Contact Header with the local IP of the other interface breaking audio. Is there any way to completely bind all IP Addresses within headers

[asterisk-users] Solved: Re: Asterisk 13.18.3 PJSIP. Wrong Port in Contact Header in Reply to REGISTER?

Hi Joshua Thank you for your reply. Indeed, Ubuntu only ships with this old version. Upgraded to 16.2. via PPA. Problem persisted. Well, I already mentioned that this is a machine with two physical interfaces with different routes which on the 'external' side handles SIP customer registrations

[asterisk-users] Asterisk 13.18.3 PJSIP. Wrong Port in Contact Header in Reply to REGISTER?

Hi List I have been pondering over a problem to use an asterisk server behind an SBC unable to successfully handle registrations. Now I observed something strange which I suspect might be a bug on the asterisk side. The SBC originates is register from Port 6011 to Port 5060 on the Asterisk.

[asterisk-users] SIP via TCP - new TCP session per call or use same session for multiple calls?

Hi List I wonder how SIP via TCP is supposed to work. Not realy Asterisk related, but I hope you experts might be able to help out :-) One of our customers has a SIP device registering via a complex NAT. To benefit from TCP Connection Tracking, he choose TCP instead of UDP. So he expected, that

Re: [asterisk-users] pjsip: How is asterisk choosing the IP address to put in the Contact header?

Short update... After some more research I found: https://community.asterisk.org/t/box-with-2-interfaces-wrong-one-chosen-in-contact-header/74705/3 And some more similar ones describing the same problem with chan_sip and pjsip. I attempted to set: external_signaling_address on my transports.

[asterisk-users] pjsip: How is asterisk choosing the IP address to put in the Contact header?

Hi Gang Server, two interfaces, routing to two different networks. Two transports defined, each bound to the corresponding ip assigned to the interface. But still, especially when an 183 message is sent, the Contact header does contain the wrong IP Address. Is this a known issue 13.18.3? Or is

Re: [asterisk-users] PJSIP device_state_busy_at, how does this work?

Hi Joshua > The option strictly controls device state. Any enforcement of a limit for > calling does not exist, and is up to you to do using various methods. > Device state could be queried and used, or GROUP[1] and GROUP_COUNT[2]. > > [1]

[asterisk-users] PJSIP device_state_busy_at, how does this work?

Hi Gang According to: https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Configuration_res_pjsip#Asterisk12Configuration_res_pjsip-endpoint_device_state_busy_at And endpoint should return busy if this number is reached. We have PBX Trunks registering to the Asterisk. So we want to limit

[asterisk-users] Avoid transcoding if both ends support same coded

Hi Gang I offer: g722 g711a g711a is mandatory. g722 is becoming more and more popular. Now if a call originates from a device which support g722 and ends on a device which does not. I see that asterisk is transcoding between g722 and g711a despite both ends supporting g711a. Google tells me,

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

[asterisk-users] bug in pjsip trust_id_outpound?

Hi Team I'm still struggling to get privacy settings passed on correctly. The Asterisk is sitting between customers and IC trunks. On the customer face, of course I have those settings: trust_id_inbound=yes trust_id_outbound=no This ensures that presentation is set to probibited, if the

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

[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] 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] 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

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

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 __

[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

[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 =>

[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

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] 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] 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] 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

[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:

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

[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] 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] 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] 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) = $_;

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

[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,

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] 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

[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

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

[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

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: 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] 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] 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] 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] 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

[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:

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] 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 =>

[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

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

han_sip reported OK to this and chan_pjsip replies with 404. Or is pjsip more intelligent and trying to prevent the phone from subscribing to itself? -Benoit- -- _ -- Bandwidth and Colocation Provided by http://www.api-digita

[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

[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] 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] 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] 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] 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] 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] 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] 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

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] 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

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

is is +41315995003 (reason no-answer;privacy=off;counter=1) From what I see in the source of chan_sip the variable ${SIPDIVERSIONREASON} should be set, but it is empty... Also ${PRIDIVERSIONREASON} is empty... I'm using: Asterisk 1.6.2.5-0ubuntu1.3 Any hints? -Benoit

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

Re: [asterisk-users] Assigning an extension to a roaming phone

to be dialed by an unknown party this way) Regards, benoit -- _ -- 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] 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] 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

[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

Re: [asterisk-users] Ekiga can register but not my IP phone

Thanks for your help. you were right it also work without a stun server adding to sip.conf: externip=78.47.x.x ; in [general] the IP of the dedicated server nat=yes ; in the description of my peer 2010/11/19, Alejandro Imass a...@p2ee.org: On Fri, Nov 19, 2010 at 8:23 AM, Benoit Chabrier c

[asterisk-users] Ekiga can register but not my IP phone

Hello, I have a Sip phone (Siemens C470IP) which works perfectly with different VoIP providers (iptel, betamax, ovh...). It also worked well with my testing server (ubuntu and inside the LAN). But now the problem i have is that the hardphone doesn't connect to my dedicated server (debian lenny /

Re: [asterisk-users] Ekiga can register but not my IP phone

Thanks Alejandro, you were right it was just a NAT problem ! i add a stun server in the phone configuration and it works :) 2010/11/19, Alejandro Imass a...@p2ee.org: On Fri, Nov 19, 2010 at 7:28 AM, Benoit Chabrier c...@chab.info wrote: Hello, I have a Sip phone (Siemens C470IP) which works

[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] 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

Re: [asterisk-users] Big practical systems

On 07/11/2010 19:29, Cary Fitch wrote: I don't want to start the How many calls can Asterisk handle? discussion or How many angels can stand on the point of a pin? discussion either. But can anyone contribute some practical knowledge of systems that take in channel bank T1s or DS3s from far

Re: [asterisk-users] Asterisk died without any message, segfault

Le 28/10/2010 08:41, Krzysztof Urbaniak a écrit : 2010/10/27 Benoitmaver...@maverick.eu.org: On 27/10/2010 12:59, Krzysztof Urbaniak wrote: Hi! We've experienced asterisk has gone without any message, it wasn't any segfault, anything in asterisk messages log that says about shutting down.

Re: [asterisk-users] Asterisk died without any message, segfault

On 27/10/2010 12:59, Krzysztof Urbaniak wrote: Hi! We've experienced asterisk has gone without any message, it wasn't any segfault, anything in asterisk messages log that says about shutting down. How do you launch asterisk ? did you try without 'safe_asterisk' or anything like it, just

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