Re: [asterisk-users] Incoming SIP call, outgoing SIP registration. PJSIP.
23.04.2019 0:27, Joshua C. Colp wrote: On Mon, Apr 22, 2019, at 2:10 PM, Pavel wrote: Tried already. "line" is good, but not perfect. Every time I restart asterisk, it will generate new random string for ";line=". So, every time I restart asterisk, registrar (Server1) will save one more contact in it's database. Some will remove obsolete contacts, but some will not. For example, FreePBX will not remove obsolete contacts, if max_contacts specified (FreePBX will set rewrite_contact=no in this case). So, after a number of Asterisk restarts, FreePBX will reject new registrations, as max_contacts is reached. It should specify remove_existing to remove old ones to make room for the new ones. That would be a FreePBX thing, though. FreePBX is an example, where it can be a critical problem. 3cx will work, but if you will restart asterisk 10 times - you will see 10 times more contacts in 3cx. When you will make call from 3cx - it will make 10 calls (10 contacts), untill they will obsolete... Unfortunately, "line" does not save random between restarts. It's also unable to specify "random" value in pjsip.conf. I'm thinking to patch res_pjsip_outbound_registration to add this feature. Am I wrong and there is another way ? I don't see any reason why this couldn't be an option. For flexibility. Not to register new fake contacts in peer PBX. It's also a security hole, as anybody can generate INVITE with ";line=random" from any IP address ! You can use an ACL to limit the endpoint to certain source IP addresses. 5+ ! Thank you, ACL is a good idea ! res_pjsip_outbound_registration will only match "line", but will not take care about source IP, ... Is there any more clear way to identify incoming INVITE/OPTIONS packets ? Not very familliar with SIP, not sure, how should it be done. There is no real defined mechanism within SIP to do this. Phones employ different mechanisms to differentiate. Some may use a similar mechanism to the line option. Some run multiple SIP transports on different ports for each account so they can differentiate based on where it came in on. Some look at the request URI coming in. Some just don't care. Sniffered some time ago how it's done in phonerlite, jitsi, linksys, ... Some use different port, some use ";rinstance=", the same like ";line=" in asterisk. Was not sure it's a right way to go. I will probably extend "line" a bit to specify it's value in pjsip.conf . It will be less than 10 lines of code. Thank you very much ! Your help will simplify my life a lot :-) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming SIP call, outgoing SIP registration. PJSIP.
On Mon, Apr 22, 2019, at 2:10 PM, Pavel wrote: > Tried already. > > "line" is good, but not perfect. > > Every time I restart asterisk, it will generate new random string for > ";line=". > > So, every time I restart asterisk, registrar (Server1) will save one > more contact in it's database. > > Some will remove obsolete contacts, but some will not. > > For example, FreePBX will not remove obsolete contacts, if max_contacts > specified (FreePBX will set rewrite_contact=no in this case). > > So, after a number of Asterisk restarts, FreePBX will reject new > registrations, as max_contacts is reached. It should specify remove_existing to remove old ones to make room for the new ones. That would be a FreePBX thing, though. > Unfortunately, "line" does not save random between restarts. > > It's also unable to specify "random" value in pjsip.conf. > > > I'm thinking to patch res_pjsip_outbound_registration to add this feature. > > Am I wrong and there is another way ? I don't see any reason why this couldn't be an option. > > It's also a security hole, as anybody can generate INVITE with > ";line=random" from any IP address ! You can use an ACL to limit the endpoint to certain source IP addresses. > > res_pjsip_outbound_registration will only match "line", but will not > take care about source IP, ... > > > > Is there any more clear way to identify incoming INVITE/OPTIONS packets ? > > Not very familliar with SIP, not sure, how should it be done. There is no real defined mechanism within SIP to do this. Phones employ different mechanisms to differentiate. Some may use a similar mechanism to the line option. Some run multiple SIP transports on different ports for each account so they can differentiate based on where it came in on. Some look at the request URI coming in. Some just don't care. -- Joshua C. Colp Digium - A Sangoma Company | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming SIP call, outgoing SIP registration. PJSIP.
Hi, Thank for your answer. 22.04.2019 23:47, Joshua C. Colp пишет: On Mon, Apr 22, 2019, at 1:43 PM, Pavel wrote: Hi, Got problems with incoming SIP calls. Scenario: Server1: 3cx or any other server Server2: Asterisk 16.2.1 . PJPROJECT 2.8 Server2 registers on Server1 with SIP ID 1121. Registration is OK. Server2 outgoing calls are OK. INVITE, unauthorized, INVITE with password, OK, RINGING,... Troubles with incoming calls / incoming INVITE's . I can not identify endpoint by IP, I have multiple registrations on the same Server1. As far as I understood, res_pjsip_endpoint_identifier_user match endpoint by "From" header, so it will not match also. match_headers also seems useless (not able to match "INVITE" string, just headers like "TO:"). Is there any way to match incoming INVITE calls ? (also OPTIONS, NOTIFY, ... packets) It should be a typical scenario, but it does not work... Is there any way to make it working ? Outbound registration provides the line option[1] which can be used to differentiate traffic in regards to different outbound registrations. It requires the remote server to adhere to the SIP RFC and report back some data we give in our Contact, so you have to test it and see if it works. [1] https://blogs.asterisk.org/2016/01/27/the-pjsip-outbound-registration-line-option/ Tried already. "line" is good, but not perfect. Every time I restart asterisk, it will generate new random string for ";line=". So, every time I restart asterisk, registrar (Server1) will save one more contact in it's database. Some will remove obsolete contacts, but some will not. For example, FreePBX will not remove obsolete contacts, if max_contacts specified (FreePBX will set rewrite_contact=no in this case). So, after a number of Asterisk restarts, FreePBX will reject new registrations, as max_contacts is reached. Unfortunately, "line" does not save random between restarts. It's also unable to specify "random" value in pjsip.conf. I'm thinking to patch res_pjsip_outbound_registration to add this feature. Am I wrong and there is another way ? It's also a security hole, as anybody can generate INVITE with ";line=random" from any IP address ! res_pjsip_outbound_registration will only match "line", but will not take care about source IP, ... Is there any more clear way to identify incoming INVITE/OPTIONS packets ? Not very familliar with SIP, not sure, how should it be done. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming SIP call, outgoing SIP registration. PJSIP.
On Mon, Apr 22, 2019, at 1:43 PM, Pavel wrote: > Hi, > > Got problems with incoming SIP calls. > > Scenario: > > Server1: 3cx or any other server > > Server2: Asterisk 16.2.1 . PJPROJECT 2.8 > > Server2 registers on Server1 with SIP ID 1121. > > Registration is OK. > > Server2 outgoing calls are OK. > > INVITE, unauthorized, INVITE with password, OK, RINGING,... > > Troubles with incoming calls / incoming INVITE's . > > I can not identify endpoint by IP, I have multiple registrations on the > same Server1. > > As far as I understood, res_pjsip_endpoint_identifier_user match > endpoint by "From" header, so it will not match also. > > match_headers also seems useless (not able to match "INVITE" string, > just headers like "TO:"). > > Is there any way to match incoming INVITE calls ? (also OPTIONS, NOTIFY, > ... packets) > > It should be a typical scenario, but it does not work... > > Is there any way to make it working ? Outbound registration provides the line option[1] which can be used to differentiate traffic in regards to different outbound registrations. It requires the remote server to adhere to the SIP RFC and report back some data we give in our Contact, so you have to test it and see if it works. [1] https://blogs.asterisk.org/2016/01/27/the-pjsip-outbound-registration-line-option/ -- Joshua C. Colp Digium - A Sangoma Company | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Incoming SIP call, outgoing SIP registration. PJSIP.
Hi, Got problems with incoming SIP calls. Scenario: Server1: 3cx or any other server Server2: Asterisk 16.2.1 . PJPROJECT 2.8 Server2 registers on Server1 with SIP ID 1121. Registration is OK. Server2 outgoing calls are OK. INVITE, unauthorized, INVITE with password, OK, RINGING,... Troubles with incoming calls / incoming INVITE's . I can not identify endpoint by IP, I have multiple registrations on the same Server1. As far as I understood, res_pjsip_endpoint_identifier_user match endpoint by "From" header, so it will not match also. match_headers also seems useless (not able to match "INVITE" string, just headers like "TO:"). Is there any way to match incoming INVITE calls ? (also OPTIONS, NOTIFY, ... packets) It should be a typical scenario, but it does not work... Is there any way to make it working ? [0.0.0.0-udp] type=transport protocol=udp bind=0.0.0.0:5060 [endpoint0](!) type=endpoint transport=0.0.0.0-udp disallow=all allow=alaw allow=ulaw t38_udptl=no t38_udptl_ec=none fax_detect=no t38_udptl_nat=no dtmf_mode=auto direct_media=yes from_domain=172.16.25.23 timers_sess_expires=1800 tone_zone=ru language=ru rewrite_contact=yes rtp_symmetric=yes force_rport=yes [registration0](!) type=registration transport=0.0.0.0-udp retry_interval=60 max_retries=10 expiration=3600 auth_rejection_permanent=yes server_uri=sip:172.16.25.23 [fxs17](endpoint0) context=from-sip-fxs aors=fxs17 outbound_auth=fxs17 from_user=1121 set_var=DAHDICHAN=17 [fxs17] type=aor qualify_frequency=60 contact=sip:1121@172.16.25.23 [fxs17] type=auth auth_type=userpass password=11 username=1121 [fxs17](registration0) outbound_auth=fxs17 client_uri=sip:1121@172.16.25.23 contact_user=fxs17 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] incoming call label
Hello Thelma, Friday, February 16, 2018, 2:16:02 AM, you wrote: > Contact: "sip:pstn-" > And it found in sip.conf only: > Found peer 'pstn-9998' for '7804715665' from 10.10.0.8:5060 > Is perhaps the name effected by the special character "-" (dash) that is > why it only matches "pstn" and take the first one it found. Will it > make a difference if I rename the port to pstn_ in configuration files. If the type=friend then it matches on IP Address and Port Number, not the user name. It will then use the first entry in the sip.conf - it does not take any notice of the name. If you change the order that the two entries appear, all the calls will appear to come from [pstn-9998] even if they come from [pstn-]. I used to set user=peer, which solved the problem for me, but I now direct all the calls to a single context in extensions.conf and then send them to their own contexts based on the DNID. -- Best regards, Julianmailto:jb_s...@trink.co.uk -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] incoming call label
Le 16/02/2018 à 05:30, the...@sys-concept.com a écrit : On 02/15/2018 04:49 PM, Joshua Colp wrote: On Thu, Feb 15, 2018, at 7:46 PM, the...@sys-concept.com wrote: Thanks again for the hint. Here is the output from asterisk. The call is coming on Audocodes gateway from: pstn- But asterisk display: Found peer 'pstn-9998' for '7804715665' from 10.10.0.8:5060 Why not loolking up "pstn-" in sip.conf? It found pstn- using 10.10.0.8:5060 - if the request always comes from the same IP address and port it has no other way built in to differentiate between the two except by matching based on username in the 'From' header. It didn't find "pstn- using 10.10.0.8:5060" The call came IN from PSTN line on audiocodes equipment to FXO port that is labelled "pstn-" so asterisk reported as such. And I think asterisk suppose to lookup this label in sip.conf to the registered entry but instead selected pstn-9998 entry; I don't know why. If the call came IN on pstn- and sip.conf has two entries: [pstn-] [pstn-9998] Why it can not distinguish between the two of them correctly? -- Thelma If your device supports SIP authentication, you can try to turn on the "match_auth_username" parameter in sip.conf. It is said to be experimental but has always worked well for me. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] incoming call label
On 02/15/2018 04:49 PM, Joshua Colp wrote: > On Thu, Feb 15, 2018, at 7:46 PM, the...@sys-concept.com wrote: > > > >> >> Thanks again for the hint. >> Here is the output from asterisk. >> >> The call is coming on Audocodes gateway from: pstn- >> >> But asterisk display: >> Found peer 'pstn-9998' for '7804715665' from 10.10.0.8:5060 >> >> Why not loolking up "pstn-" in sip.conf? > > It found pstn- using 10.10.0.8:5060 - if the request always comes from > the same IP address and port it has no other way built in to differentiate > between the two except by matching based on username in the 'From' header. It didn't find "pstn- using 10.10.0.8:5060" The call came IN from PSTN line on audiocodes equipment to FXO port that is labelled "pstn-" so asterisk reported as such. And I think asterisk suppose to lookup this label in sip.conf to the registered entry but instead selected pstn-9998 entry; I don't know why. If the call came IN on pstn- and sip.conf has two entries: [pstn-] [pstn-9998] Why it can not distinguish between the two of them correctly? -- Thelma -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] incoming call label
Thelma On 02/15/2018 07:16 PM, the...@sys-concept.com wrote: > > On 02/15/2018 04:49 PM, Joshua Colp wrote: >> On Thu, Feb 15, 2018, at 7:46 PM, the...@sys-concept.com wrote: >> >> >> >>> >>> Thanks again for the hint. >>> Here is the output from asterisk. >>> >>> The call is coming on Audocodes gateway from: pstn- >>> >>> But asterisk display: >>> Found peer 'pstn-9998' for '7804715665' from 10.10.0.8:5060 >>> >>> Why not loolking up "pstn-" in sip.conf? >> >> It found pstn- using 10.10.0.8:5060 - if the request always comes from >> the same IP address and port it has no other way built in to differentiate >> between the two except by matching based on username in the 'From' header. >> > > Call comes from same IP address always. > To comes form Audiocode: > > <--- SIP read from UDP:10.10.0.8:5060 ---> > INVITE sip:4@10.10.0.4 SIP/2.0 > Via: SIP/2.0/UDP 10.10.0.8;branch=z9hG4bKac766808844 > Max-Forwards: 70 > From: "Z" ;tag=1c766802762 > To: > Call-ID: 7668022781522018162620@10.10.0.8 > CSeq: 1 INVITE > Contact: > > Contact: "sip:pstn-" > > And it found in sip.conf only: > Found peer 'pstn-9998' for '7804715665' from 10.10.0.8:5060 > > Is perhaps the name effected by the special character "-" (dash) that is > why it only matches "pstn" and take the first one it found. Will it > make a difference if I rename the port to pstn_ in configuration files. > > -- > Thelma sip show peers Name/username HostDyn Forcerport ComediaACL Port Status Description pstn-/voice- 10.10.0.8D No No 5060 Unmonitored pstn-9998/fax-999810.10.0.8D No No 5060 Unmonitored -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] incoming call label
On 02/15/2018 04:49 PM, Joshua Colp wrote: > On Thu, Feb 15, 2018, at 7:46 PM, the...@sys-concept.com wrote: > > > >> >> Thanks again for the hint. >> Here is the output from asterisk. >> >> The call is coming on Audocodes gateway from: pstn- >> >> But asterisk display: >> Found peer 'pstn-9998' for '7804715665' from 10.10.0.8:5060 >> >> Why not loolking up "pstn-" in sip.conf? > > It found pstn- using 10.10.0.8:5060 - if the request always comes from > the same IP address and port it has no other way built in to differentiate > between the two except by matching based on username in the 'From' header. > Call comes from same IP address always. To comes form Audiocode: <--- SIP read from UDP:10.10.0.8:5060 ---> INVITE sip:4@10.10.0.4 SIP/2.0 Via: SIP/2.0/UDP 10.10.0.8;branch=z9hG4bKac766808844 Max-Forwards: 70 From: "Z" ;tag=1c766802762 To: Call-ID: 7668022781522018162620@10.10.0.8 CSeq: 1 INVITE Contact: Contact: "sip:pstn-" And it found in sip.conf only: Found peer 'pstn-9998' for '7804715665' from 10.10.0.8:5060 Is perhaps the name effected by the special character "-" (dash) that is why it only matches "pstn" and take the first one it found. Will it make a difference if I rename the port to pstn_ in configuration files. -- Thelma -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] incoming call label
On Thu, Feb 15, 2018, at 7:46 PM, the...@sys-concept.com wrote: > > Thanks again for the hint. > Here is the output from asterisk. > > The call is coming on Audocodes gateway from: pstn- > > But asterisk display: > Found peer 'pstn-9998' for '7804715665' from 10.10.0.8:5060 > > Why not loolking up "pstn-" in sip.conf? It found pstn- using 10.10.0.8:5060 - if the request always comes from the same IP address and port it has no other way built in to differentiate between the two except by matching based on username in the 'From' header. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] incoming call label
On 02/15/2018 04:08 PM, Joshua Colp wrote: > On Thu, Feb 15, 2018, at 7:03 PM, the...@sys-concept.com wrote: >> On 02/15/2018 03:44 PM, Joshua Colp wrote: >>> On Thu, Feb 15, 2018, at 6:43 PM, the...@sys-concept.com wrote: I'm using Audio-codes MP-114 unit and it has two public lines PSTN ports IN audocodes setting I have: "EndPoint Phone Number" Channel: 3phone number: pstn- Channel: 4phone number: pstn-9998 When I am calling " pstn-" the port number "Channel:3" lights up but asterisk is showing that the call is coming on "pstn-9998" -- Executing . Answer("SIP/pstn-9998 Asterisk should be showing "pstn-" (not pstn-9998) Where is this label coming from? >>> >>> It is from the SIP entry in sip.conf that it was matched against. >>> >> >> Thanks for the input. >> >> In sip.conf I have relevant entries. >> >> [pstn-] ; incoming/outgoing calls on FXO port >> type=friend >> secret=spa354 >> username=voice- >> mailbox=622 ; just for audiocodes error complain >> host=dynamic >> canreinvite=no ; (dtmf not wroking correctly without this one) >> disallow=all >> allow=ulaw >> allow=alaw >> nat=no >> context=incoming >> callgroup=1 >> pickupgroup=1 >> insecure=invite >> >> [pstn-9998] >> type=friend >> secret=158567 >> username=fax-9998 >> insecure=invite >> mailbox=622 ; just for audiocodes error complain >> host=dynamic >> canreinvite=no ; (dtmf not wroking correctly without this one) >> disallow=all >> allow=ulaw >> allow=alaw >> nat=no >> context=incoming >> callgroup=1 >> pickupgroup= >> >> My asterisk registration is correct as well: >> sip show users >> Username Secret Accountcode Def.Context >> ACL Forcerport >> pstn-9998 158567 incoming >> No No >> pstn- spa354 incoming >> No No >> >> Caller display ID from PSTN on FXO ports are working OK. >> The [pstn-] is channel: 4 >> The [pstn-9998] is channel: 3 >> >> If the call on Audocode is lighting UP "channel:3" the sip.conf should >> associate that call with [pstn-] (and not [pstn-9998]) > > Not necessarily. You appear to be doing IP+port based matching. If requests > always come from the same source IP address and port, then it would match > only one. Turning on sip debug using "sip set debug on" and verbosity using > "core set debug 9" would give you more information about each packet > (including where it is from) and what was actually matched based on it. Thanks again for the hint. Here is the output from asterisk. The call is coming on Audocodes gateway from: pstn- But asterisk display: Found peer 'pstn-9998' for '7804715665' from 10.10.0.8:5060 Why not loolking up "pstn-" in sip.conf? <--- SIP read from UDP:10.10.0.8:5060 ---> INVITE sip:4@10.10.0.4 SIP/2.0 Via: SIP/2.0/UDP 10.10.0.8;branch=z9hG4bKac766808844 Max-Forwards: 70 From: "Z" ;tag=1c766802762 To: Call-ID: 7668022781522018162620@10.10.0.8 CSeq: 1 INVITE Contact: Supported: em,100rel,timer,replaces,path,early-session,resource-priority,sdp-anat Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE User-Agent: Audiocodes-Sip-Gateway-/v.5.80A.032.003 Content-Type: application/sdp Content-Disposition: session Content-Length: 249 v=0 o=AudiocodesGW 766797875 766797759 IN IP4 10.10.0.8 s=Phone-Call c=IN IP4 10.10.0.8 t=0 0 m=audio 6000 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-> --- (14 headers 12 lines) --- Sending to 10.10.0.8:5060 (no NAT) Sending to 10.10.0.8:5060 (no NAT) Using INVITE request as basis request - 7668022781522018162620@10.10.0.8 Found peer 'pstn-9998' for '7804715665' from 10.10.0.8:5060 == Using SIP RTP CoS mark 5 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 101 Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Found audio description format telephone-event for ID 101 Capabilities: us - (ulaw|alaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 10.10.0.8:6000 Looking for 4 in incoming (domain 10.10.0.4) list_route: hop: -- Thelma -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] incoming call label
On Thu, Feb 15, 2018, at 7:03 PM, the...@sys-concept.com wrote: > On 02/15/2018 03:44 PM, Joshua Colp wrote: > > On Thu, Feb 15, 2018, at 6:43 PM, the...@sys-concept.com wrote: > >> I'm using Audio-codes MP-114 unit and it has two public lines PSTN ports > >> > >> IN audocodes setting I have: > >> "EndPoint Phone Number" > >> > >> Channel: 3phone number: pstn- > >> Channel: 4phone number: pstn-9998 > >> > >> When I am calling " pstn-" the port number "Channel:3" lights up but > >> asterisk is showing that the call is coming on "pstn-9998" > >> > >> -- Executing . Answer("SIP/pstn-9998 > >> > >> Asterisk should be showing "pstn-" (not pstn-9998) > >> Where is this label coming from? > > > > It is from the SIP entry in sip.conf that it was matched against. > > > > Thanks for the input. > > In sip.conf I have relevant entries. > > [pstn-] ; incoming/outgoing calls on FXO port > type=friend > secret=spa354 > username=voice- > mailbox=622 ; just for audiocodes error complain > host=dynamic > canreinvite=no ; (dtmf not wroking correctly without this one) > disallow=all > allow=ulaw > allow=alaw > nat=no > context=incoming > callgroup=1 > pickupgroup=1 > insecure=invite > > [pstn-9998] > type=friend > secret=158567 > username=fax-9998 > insecure=invite > mailbox=622 ; just for audiocodes error complain > host=dynamic > canreinvite=no ; (dtmf not wroking correctly without this one) > disallow=all > allow=ulaw > allow=alaw > nat=no > context=incoming > callgroup=1 > pickupgroup= > > My asterisk registration is correct as well: > sip show users > Username Secret Accountcode Def.Context > ACL Forcerport > pstn-9998 158567 incoming > No No > pstn- spa354 incoming > No No > > Caller display ID from PSTN on FXO ports are working OK. > The [pstn-] is channel: 4 > The [pstn-9998] is channel: 3 > > If the call on Audocode is lighting UP "channel:3" the sip.conf should > associate that call with [pstn-] (and not [pstn-9998]) Not necessarily. You appear to be doing IP+port based matching. If requests always come from the same source IP address and port, then it would match only one. Turning on sip debug using "sip set debug on" and verbosity using "core set debug 9" would give you more information about each packet (including where it is from) and what was actually matched based on it. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] incoming call label
On 02/15/2018 03:44 PM, Joshua Colp wrote: > On Thu, Feb 15, 2018, at 6:43 PM, the...@sys-concept.com wrote: >> I'm using Audio-codes MP-114 unit and it has two public lines PSTN ports >> >> IN audocodes setting I have: >> "EndPoint Phone Number" >> >> Channel: 3phone number: pstn- >> Channel: 4phone number: pstn-9998 >> >> When I am calling " pstn-" the port number "Channel:3" lights up but >> asterisk is showing that the call is coming on "pstn-9998" >> >> -- Executing . Answer("SIP/pstn-9998 >> >> Asterisk should be showing "pstn-" (not pstn-9998) >> Where is this label coming from? > > It is from the SIP entry in sip.conf that it was matched against. > Thanks for the input. In sip.conf I have relevant entries. [pstn-] ; incoming/outgoing calls on FXO port type=friend secret=spa354 username=voice- mailbox=622 ; just for audiocodes error complain host=dynamic canreinvite=no ; (dtmf not wroking correctly without this one) disallow=all allow=ulaw allow=alaw nat=no context=incoming callgroup=1 pickupgroup=1 insecure=invite [pstn-9998] type=friend secret=158567 username=fax-9998 insecure=invite mailbox=622 ; just for audiocodes error complain host=dynamic canreinvite=no ; (dtmf not wroking correctly without this one) disallow=all allow=ulaw allow=alaw nat=no context=incoming callgroup=1 pickupgroup= My asterisk registration is correct as well: sip show users Username Secret Accountcode Def.Context ACL Forcerport pstn-9998 158567 incoming No No pstn- spa354 incoming No No Caller display ID from PSTN on FXO ports are working OK. The [pstn-] is channel: 4 The [pstn-9998] is channel: 3 If the call on Audocode is lighting UP "channel:3" the sip.conf should associate that call with [pstn-] (and not [pstn-9998]) -- Thelma -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] incoming call label
On Thu, Feb 15, 2018, at 6:43 PM, the...@sys-concept.com wrote: > I'm using Audio-codes MP-114 unit and it has two public lines PSTN ports > > IN audocodes setting I have: > "EndPoint Phone Number" > > Channel: 3phone number: pstn- > Channel: 4phone number: pstn-9998 > > When I am calling " pstn-" the port number "Channel:3" lights up but > asterisk is showing that the call is coming on "pstn-9998" > > -- Executing . Answer("SIP/pstn-9998 > > Asterisk should be showing "pstn-" (not pstn-9998) > Where is this label coming from? It is from the SIP entry in sip.conf that it was matched against. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] incoming call label
I'm using Audio-codes MP-114 unit and it has two public lines PSTN ports IN audocodes setting I have: "EndPoint Phone Number" Channel: 3phone number: pstn- Channel: 4phone number: pstn-9998 When I am calling " pstn-" the port number "Channel:3" lights up but asterisk is showing that the call is coming on "pstn-9998" -- Executing . Answer("SIP/pstn-9998 Asterisk should be showing "pstn-" (not pstn-9998) Where is this label coming from? -- Thelma -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming Call by DID
On Wednesday 26 Oct 2016, KyD wrote: > Hi, > > My sip provider gave me 2 numbers for the incoming call via pstn. > > nro1 = 12341234 > nro2 = 45674567 > > I have a dialplan for each. > if i put this on my dialplan: > > exten => s,1,Dial(SIP/1001) > exten => Hangup() > > Works! > > But if i put one of them: > > exten => 12341234,1,Dial(SIP/1001) > exten => _1234,1,Dial(SIP/1001) > exten => 45674567,1,Dial(SIP/1001) > exten => _4567,1,Dial(SIP/1001) > > incoming calls do not arrive. > > Any ideas? The incoming call must be arriving with ${EXTEN} containing something that doesn't match 12341234, _1234, 45674567 or _4567, so it is not triggering any of the extensions in your dialplan. Maybe it still has the STD code or even the IDD code prepended. (Been caught this way once before . our old ISDN-30 provider used to send just the local number, then we moved to a new ISDN-30 provider who send the number with STD code but no initial 0. Cue frantic editing of dialplan before rest of staff arrived .) So try this; exten => s,1,NoOp(Incoming call for '${EXTEN}') exten => s,n,Dial(SIP/1001) exten => s,n,Hangup() Run `# asterisk -vvvr`, dial one of your DDI numbers from a mobile phone and watch the messages scrolling past. Now you will be seeing exactly what ${EXTEN} contains when a call comes in, so you should be able to work out what is going on, and craft your extension expressions to suit. If in doubt, post an excerpt from your CLI output. -- AJS Note: Originating address only accepts e-mail from list! If replying off- list, change address to asterisk1list at earthshod dot co dot uk . -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming Call by DID
It seems like your SIP provider is not sending and DID information, or that the information is not being sent in the same format you are using in your dialplan. You can check this by looking at the SIP debug information for the inbound calls and/or by checking with your SIP provider (that they are sending the DID number and what format it is in). All the best, David On 27 Oct 2016 5:21 am, "KyD" wrote: Hi, My sip provider gave me 2 numbers for the incoming call via pstn. nro1 = 12341234 nro2 = 45674567 I have a dialplan for each. if i put this on my dialplan: exten => s,1,Dial(SIP/1001) exten => Hangup() Works! But if i put one of them: exten => 12341234,1,Dial(SIP/1001) exten => _1234,1,Dial(SIP/1001) exten => 45674567,1,Dial(SIP/1001) exten => _4567,1,Dial(SIP/1001) incoming calls do not arrive. Any ideas? -- KyD GNU/Linux SysAdmin Quanto mais você sabe, mais você percebe que você não sabe nada. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk. org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Incoming Call by DID
Hi, My sip provider gave me 2 numbers for the incoming call via pstn. nro1 = 12341234 nro2 = 45674567 I have a dialplan for each. if i put this on my dialplan: exten => s,1,Dial(SIP/1001) exten => Hangup() Works! But if i put one of them: exten => 12341234,1,Dial(SIP/1001) exten => _1234,1,Dial(SIP/1001) exten => 45674567,1,Dial(SIP/1001) exten => _4567,1,Dial(SIP/1001) incoming calls do not arrive. Any ideas? -- KyD GNU/Linux SysAdmin Quanto mais você sabe, mais você percebe que você não sabe nada. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming calls from Andrews & Arnold failing to authenticate
Hello Phil, On Saturday, April 23, 2016, 11:11:29 PM, you wrote: > Actually, this is now sorted. It turns out the latest recommended > configs on the A&A wiki had peer vs. user confusion. On correcting > this, all was well. I'm glad you found it. It look me a while to track down that problem when I had it. The one that was hardest for me to track down was a slight mis-match between the RTP ports in Asterisk and the corresponding ports open on a firewall, which resulted in about 1 in 10 calls having no audio! Doh! -- Best regards, Julianmailto:jb_s...@trink.co.uk -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming calls from Andrews & Arnold failing to authenticate
On Sat, 23 Apr 2016 22:45:32 +0100 Julian Beach wrote: > Hello Phil, > > I have a couple of lines with A&A, and I have not been having any > problems recently. When I have had similar problems in the past, it > has been an issue with the SIP config. I originally had a number of > contexts set up in sip.conf to handle the lines coming in (such as > [aa-line1], [aa-line2]) each with their own username and password > settings. The type=user setting was critical, because all the calls > came from the same IP address, and using type=peer caused matching > problems which resulted in authentication failures. This got too > complex to manage once I added in all the IP addresses A&A calls might > come in from. so I simplified the setup. > > I now have just one context in sip.conf to handle incoming A&A calls, > with the same username for all lines, and type=peer. Calls are then > sent to extensions.conf, where the calls are directed to the correct > call-handler for the line based on the CID. Here is the setup in > sip.conf for A&A calls: Actually, this is now sorted. It turns out the latest recommended configs on the A&A wiki had peer vs. user confusion. On correcting this, all was well. -- Phil Reynolds mail: phil-aster...@tinsleyviaduct.com Web: http://phil.tinsleyviaduct.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming calls from Andrews & Arnold failing to authenticate
Hello Phil, On Saturday, April 23, 2016, 12:19:15 PM, you wrote: > I have checked that the username and password in my config agree both > ends, and have even tried changing them. > The bulk of my calls come in on A&A, so I am obviously trying to find > out what has gone wrong. No-one else is seeing any problem. What do I > need to do to track this down? I have a couple of lines with A&A, and I have not been having any problems recently. When I have had similar problems in the past, it has been an issue with the SIP config. I originally had a number of contexts set up in sip.conf to handle the lines coming in (such as [aa-line1], [aa-line2]) each with their own username and password settings. The type=user setting was critical, because all the calls came from the same IP address, and using type=peer caused matching problems which resulted in authentication failures. This got too complex to manage once I added in all the IP addresses A&A calls might come in from. so I simplified the setup. I now have just one context in sip.conf to handle incoming A&A calls, with the same username for all lines, and type=peer. Calls are then sent to extensions.conf, where the calls are directed to the correct call-handler for the line based on the CID. Here is the setup in sip.conf for A&A calls: --- sip.conf [aa-incoming](!) type=peer context=aa-incoming insecure=invite transport=udp disallow=all allow=alaw trustrpid=yes sendrpid=yes ; IPv4 hostnames [voiceless-1](aa-incoming) host=a4.voiceless.aa.net.uk [voiceless-2](aa-incoming) host=b4.voiceless.aa.net.uk [voiceless-3](aa-incoming) host=c4.voiceless.aa.net.uk [voiceless-4](aa-incoming) host=d4.voiceless.aa.net.uk [voiceless-5](aa-incoming) host=e4.voiceless.aa.net.uk [voiceless-6](aa-incoming) host=f4.voiceless.aa.net.uk [voiceless-7](aa-incoming) host=g4.voiceless.aa.net.uk [voiceless-8](aa-incoming) host=h4.voiceless.aa.net.uk [voiceless-9](aa-incoming) host=i4.voiceless.aa.net.uk [voiceless-10](aa-incoming) host=j4.voiceless.aa.net.uk --- The trustrpid and sendrpid settings were important. --- extensions.conf (DNIDs changed) === [aa-incoming] exten => 4401,1,Goto(from-aa-line1,s,1) exten => 4402,1,Goto(from-aa-line2,s,1) exten => 4403,1,Goto(from-aa-line3,s,1) --- Hope this helps. Julian -- Best regards, Julianmailto:jb_s...@trink.co.uk -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Incoming calls from Andrews & Arnold failing to authenticate
I have service with both VoIPtalk.org and Andrews & Arnold (aa.net.uk). VoIPtalk calls are unauthenticated and reach me fine, but Andrews & Arnold calls are authenticated. The last call I successfully received was on Tuesday afternoon. Initially, A&A were for some odd reason not sending calls to my server, but that has been resolved. The problem now is that the calls fail to authenticate, and are therefore rejected - error 403 is presented to them, and I see this in Asterisk's console: [Apr 23 11:53:19] NOTICE[27398][C-0004]: chan_sip.c:25535 handle_request_invite: Failed to authenticate device "X XX" ;tag=201604231153191 I have checked that the username and password in my config agree both ends, and have even tried changing them. The bulk of my calls come in on A&A, so I am obviously trying to find out what has gone wrong. No-one else is seeing any problem. What do I need to do to track this down? -- Phil Reynolds mail: phil-aster...@tinsleyviaduct.com Web: http://phil.tinsleyviaduct.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming INVITE with Portability Info and LRN
On Sun, 20 Mar 2016, Trey Hilyard wrote: On Mar 18, 2016 8:27 PM, "Steve Edwards" wrote: >> >> On Fri, 18 Mar 2016, Trey Hilyard wrote: >> >>> I thought this would be as easy as >>> exten => _XX\;rn=+1913663,1,Goto(from_pstn,${EXTEN:0:10}) > > > How about something like: > > [parse-lrn] > exten = _x.,1, verbose(1,[${EXTEN}@${CONTEXT}]) > same = n, set(DID=${CUT(EXTEN,\;,1)}) > same = n, set(LRN=${CUT(EXTEN,\;,2):3:12}) > same = n, execif($["${LRN:0:1}" = "+"]?set(LRN=${LRN:1})) > same = n, execif($["${LRN:0:1}" = "1"]?set(LRN=${LRN:1})) > same = n, goto(${LRN},${DID},1) > same = n, hangup() That's a good one. One thing it doesn't do is actually validate that the LRN is mine, but that shouldn't be tough to add now the the LRN is in its own variable. Thanks for the help! If the LRN is not yours, you will not have a matching context so the goto() will run the invalid handler (the 'i' extension). You could play an appropriate message there. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming INVITE with Portability Info and LRN
On Mar 18, 2016 8:27 PM, "Steve Edwards" wrote: >> >> On Fri, 18 Mar 2016, Trey Hilyard wrote: >> >>> I thought this would be as easy as >>> exten => _XX\;rn=+1913663,1,Goto(from_pstn,${EXTEN:0:10}) > > > How about something like: > > [parse-lrn] > exten = _x.,1, verbose(1,[${EXTEN}@${CONTEXT}]) > same = n, set(DID=${CUT(EXTEN,\;,1)}) > same = n, set(LRN=${CUT(EXTEN,\;,2):3:12}) > same = n, execif($["${LRN:0:1}" = "+"]?set(LRN=${LRN:1})) > same = n, execif($["${LRN:0:1}" = "1"]?set(LRN=${LRN:1})) > same = n, goto(${LRN},${DID},1) > same = n, hangup() That's a good one. One thing it doesn't do is actually validate that the LRN is mine, but that shouldn't be tough to add now the the LRN is in its own variable. Thanks for the help! > > -- > Thanks in advance, > - > Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST > https://www.linkedin.com/in/steve-edwards-4244281 > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming INVITE with Portability Info and LRN
On Fri, 18 Mar 2016, Trey Hilyard wrote: I thought this would be as easy as exten => _XX\;rn=+1913663,1,Goto(from_pstn,${EXTEN:0:10}) Have you tried the '_!.' pattern? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Incoming INVITE with Portability Info and LRN
I am trying to set up my Asterisk server so that it will recognize an incoming call to the Asterisk's own Location Routing Number (LRN), validating the "rn" in the INVITE and then using the Called Number from the INVITE as the extension in the dialplan. The INVITE R-URI looks like: INVITE sip:+19135041291;rn=+1913663;npdi@12.4.240.200:5060;user=phone;transport=udp SIP/2.0 The +1913663000 is the LRN of the Asterisk box, so I would want to have the dialplan validate that the "rn" is that number. The +19136631291 is the extension within the system that they are trying to reach, that extension will vary, and will have an exten defined in the dialplan. I assume that this is just going to require that I do some matching and substring-type variable replacement to hit a context with just the Called Number part of the request, but I wondered if anyone had a working example of this before I started putting too much effort into it. As a PBX, Asterisk doesn't have to worry about portability, but I am using it to simulate a full-blown Class 5 switch, so I have to have an LRN assigned to it to allow users to port to that switch. -Trey -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming INVITE with Portability Info and LRN
On Fri, Mar 18, 2016 at 10:49 AM Administrator TOOTAI wrote: > Le 18/03/2016 16:20, Trey Hilyard a écrit : > > I am trying to set up my Asterisk server so that it will recognize an > > incoming call to the Asterisk's own Location Routing Number (LRN), > > validating the "rn" in the INVITE and then using the Called Number from > > the INVITE as the extension in the dialplan. > > > > The INVITE R-URI looks like: > > INVITE > > sip:+19135041291;rn=+1913663;npdi@12.4.240.200 > :5060;user=phone;transport=udp > > SIP/2.0 > > > > The +1913663000 is the LRN of the Asterisk box, so I would want to have > > the dialplan validate that the "rn" is that number. The +19136631291 is > > the extension within the system that they are trying to reach, that > > extension will vary, and will have an exten defined in the dialplan. > > > > I assume that this is just going to require that I do some matching and > > substring-type variable replacement to hit a context with just the > > Called Number part of the request, but I wondered if anyone had a > > working example of this before I started putting too much effort into it. > > Use the SIP_HEADER function > > http://www.voip-info.org/wiki/view/Asterisk+func+sip_header I am not sure that this is needed here. The Request URI has all of the values that I need. I agree that I might need to CUT part of the R-URI, but I don't need access to any other header to find the info I need. When the call arrives at the Asterisk right now, this is the exten/context that it is hitting, so it already has the info I need: Executing [9135041291;rn=+1913663;npdi@from_pstn:1] As far as I can tell, I think that I just need to figure out how to make an extension entry that matches on the "rn=+1913663\;npdi" and then moves to another context (or same one) with ${EXTEN,0,10}. I just can't get that first extension to match on the RN value. > > > -- > Daniel > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: >http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming INVITE with Portability Info and LRN
Le 18/03/2016 16:20, Trey Hilyard a écrit : I am trying to set up my Asterisk server so that it will recognize an incoming call to the Asterisk's own Location Routing Number (LRN), validating the "rn" in the INVITE and then using the Called Number from the INVITE as the extension in the dialplan. The INVITE R-URI looks like: INVITE sip:+19135041291;rn=+1913663;npdi@12.4.240.200:5060;user=phone;transport=udp SIP/2.0 The +1913663000 is the LRN of the Asterisk box, so I would want to have the dialplan validate that the "rn" is that number. The +19136631291 is the extension within the system that they are trying to reach, that extension will vary, and will have an exten defined in the dialplan. I assume that this is just going to require that I do some matching and substring-type variable replacement to hit a context with just the Called Number part of the request, but I wondered if anyone had a working example of this before I started putting too much effort into it. Use the SIP_HEADER function http://www.voip-info.org/wiki/view/Asterisk+func+sip_header -- Daniel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming INVITE with Portability Info and LRN
I thought this would be as easy as exten => _XX\;rn=+1913663,1,Goto(from_pstn,${EXTEN:0:10}) But it appears that the pattern match doesn't work once I get to the "r" in "rn". I am assuming that the pattern match doesn't like dealing with characters without taking the entire URI. I am working on a plan using a lot more CUTs than I think I should need, but we'll see if it works. On Fri, Mar 18, 2016 at 10:58 AM Trey Hilyard wrote: > On Fri, Mar 18, 2016 at 10:49 AM Administrator TOOTAI > wrote: > >> Le 18/03/2016 16:20, Trey Hilyard a écrit : >> > I am trying to set up my Asterisk server so that it will recognize an >> > incoming call to the Asterisk's own Location Routing Number (LRN), >> > validating the "rn" in the INVITE and then using the Called Number from >> > the INVITE as the extension in the dialplan. >> > >> > The INVITE R-URI looks like: >> > INVITE >> > sip:+19135041291;rn=+1913663;npdi@12.4.240.200 >> :5060;user=phone;transport=udp >> > SIP/2.0 >> > >> > The +1913663000 is the LRN of the Asterisk box, so I would want to have >> > the dialplan validate that the "rn" is that number. The +19136631291 is >> > the extension within the system that they are trying to reach, that >> > extension will vary, and will have an exten defined in the dialplan. >> > >> > I assume that this is just going to require that I do some matching and >> > substring-type variable replacement to hit a context with just the >> > Called Number part of the request, but I wondered if anyone had a >> > working example of this before I started putting too much effort into >> it. >> >> Use the SIP_HEADER function >> >> http://www.voip-info.org/wiki/view/Asterisk+func+sip_header > > > I am not sure that this is needed here. The Request URI has all of the > values that I need. I agree that I might need to CUT part of the R-URI, but > I don't need access to any other header to find the info I need. > > When the call arrives at the Asterisk right now, this is the exten/context > that it is hitting, so it already has the info I need: > Executing [9135041291;rn=+1913663;npdi@from_pstn:1] > > As far as I can tell, I think that I just need to figure out how to make > an extension entry that matches on the "rn=+1913663\;npdi" and then > moves to another context (or same one) with ${EXTEN,0,10}. > > I just can't get that first extension to match on the RN value. > > > >> >> >> -- >> Daniel >> >> -- >> _ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >>http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >>http://lists.digium.com/mailman/listinfo/asterisk-users >> > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming INVITE with Portability Info and LRN
On Fri, 18 Mar 2016, Steve Edwards wrote: Have you tried the '_!.' pattern? The '_x.' pattern works fine. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming INVITE with Portability Info and LRN
On Fri, 18 Mar 2016, Trey Hilyard wrote: I thought this would be as easy as exten => _XX\;rn=+1913663,1,Goto(from_pstn,${EXTEN:0:10}) How about something like: [parse-lrn] exten = _x.,1, verbose(1,[${EXTEN}@${CONTEXT}]) same = n, set(DID=${CUT(EXTEN,\;,1)}) same = n, set(LRN=${CUT(EXTEN,\;,2):3:12}) same = n, execif($["${LRN:0:1}" = "+"]?set(LRN=${LRN:1})) same = n, execif($["${LRN:0:1}" = "1"]?set(LRN=${LRN:1})) same = n, goto(${LRN},${DID},1) same = n, hangup() -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming Fax Issue with Asterisk 11.7 and Digium Fax
Larry Moore omninet.net.au> writes: > > sip.conf > > [general] > faxdetect=t38 > > [sipcall.ch] > directmedia=no > > In extensions.conf change Wait(2) to Wait(5), if your VSP sends you a > T.38 re-invite this will trigger the switch to the Fax extension. > > If this proves successful you can work on removing the Wait() from your > dialplan as Asterisk will remain in the audio path and should > successfully switch to the fax extension if extension 200 or 201 answer > a call that happens to be a fax. > > Larry. > Hi to all Sorry to bump this old thread. Well, I found a while ago finally the reason why T.38 doesn't work in conjunction with Swiss VoIP provider sipcall. Despite T.38 is stated as "supported", that provider does NOT support T.38. Their T.38 gateway has some fundamental negotiation problems, - it "exceeds the T4 timer of the T.30 protocol". Therefore, T.38 faxing does not work. http://wiki.innovaphone.com/index.php?title=Howto:Sipcall_business_-_SIP_Provider_Compatibility_Test Sipcall has confirmed me that they work now on a solution. Will see... Kind regards, Clemens -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming webrtc call succeeds in Firefox but fails in Google Chrome
El 20/01/16 a las 18:33, Alex Villacís Lasso escribió: El 20/01/16 a las 16:25, Alex Villacís Lasso escribió: Partial fix: Google Chrome accepts the call if videosupport is set to "no". This is the SDP of the successful INVITE that Chrome accepts: INVITE sip:8cj802p8@192.0.2.240;transport=wss SIP/2.0 Via: SIP/2.0/WS 10.1.0.4:5060;branch=z9hG4bK65071dc5;rport Max-Forwards: 70 From: "Anonymous" ;tag=as474012b5 To: Contact: Call-ID: 73b82a5b6fbaab50741cd99424b1f31a@10.1.0.4:5060 CSeq: 102 INVITE User-Agent: FPBX-2.11.0(11.20.0) Date: Wed, 20 Jan 2016 23:27:51 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 937 v=0 o=root 2094440180 2094440180 IN IP4 10.1.0.4 s=Asterisk PBX 11.21.0 c=IN IP4 10.1.0.4 t=0 0 m=audio 18758 UDP/TLS/RTP/SAVPF 4 0 3 8 18 110 9 101 a=rtpmap:4 G723/8000 a=fmtp:4 annexa=no a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:110 speex/8000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=ice-ufrag:0033e2a20fe4becd1c34b13f5efcf1e3 a=ice-pwd:65693b30588f061710baa3584253eaba a=candidate:Ha010004 1 UDP 2130706431 10.1.0.4 18758 typ host a=candidate:Hc0a80592 1 UDP 2130706431 192.168.5.146 18758 typ host a=candidate:Ha010004 2 UDP 2130706430 10.1.0.4 18759 typ host a=candidate:Hc0a80592 2 UDP 2130706430 192.168.5.146 18759 typ host a=connection:new a=setup:actpass a=fingerprint:SHA-256 A1:9E:D0:11:26:AA:30:00:AF:06:87:9C:A7:C2:70:4F:A3:3F:89:B5:7D:5C:FA:90:89:7E:D0:8A:F0:72:F5:2A a=sendrecv However, I want to enable full video passthrough. Is this some kind of video codec incompatibility? If I enable allow=vp8 to the set of allowed codecs, Chrome accepts the video call, but now I get no sound with the demo-congrats command. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming webrtc call succeeds in Firefox but fails in Google Chrome
El 20/01/16 a las 16:25, Alex Villacís Lasso escribió: I am having trouble getting Google Chrome to accept a WebRTC call coming from Asterisk, even though Firefox can (now) accept the same call without issue. My setup is as follows: Server: CentOS 7 x86_64 (Elastix 4 RC) with IP: 10.1.0.4 192.168.5.146 asterisk-11.21.0 patched to work around https://issues.asterisk.org/jira/browse/ASTERISK-25659 openssl-1.0.1e-51.el7_2.2.x86_64 [root@elx4 ~]# openssl version OpenSSL 1.0.1e-fips 11 Feb 2013 [root@elx4 ~]# openssl ecparam -list_curves secp384r1 : NIST/SECG curve over a 384 bit prime field secp521r1 : NIST/SECG curve over a 521 bit prime field prime256v1: X9.62/SECG curve over a 256 bit prime field Client: Fedora 23 x86_64 Linphone (linphone-3.6.1-10.fc23.x86_64) Firefox 43 (firefox-43.0.3-1.fc23.x86_64) Google Chrome (google-chrome-stable-47.0.2526.111-1.x86_64) SIP.js 0.7.2 I set up my SIP configuration to have two SIP accounts. Account 1000 is the Linphone and 1001 is the webrtc: [general] faxdetect=no vmexten=*97 context=from-sip-external callerid=Unknown notifyringing=yes notifyhold=yes tos_sip=cs3 tos_audio=ef tos_video=af41 alwaysauthreject=yes useragent=FPBX-2.11.0(11.20.0) disallow=all allow=g723 allow=ulaw allow=gsm allow=alaw allow=g729 allow=speex allow=g722 allow=h264 allow=h263p allow=h263 allow=h261 tlsenable=yes tlsbindaddr=0.0.0.0 tlscipher=ALL tlsclientmethod=tlsv1 tlscertfile=/etc/asterisk/keys/asterisk.pem tlscafile=/etc/asterisk/keys/ca.crt callevents=no jbenable=no videosupport=yes allowguest=no srvlookup=no defaultexpiry=120 minexpiry=60 maxexpiry=3600 registerattempts=0 registertimeout=20 g726nonstandard=no maxcallbitrate=384 canreinvite=no rtptimeout=30 rtpholdtimeout=300 rtpkeepalive=0 checkmwi=10 notifyringing=yes notifyhold=yes nat=yes [1000] deny=0.0.0.0/0.0.0.0 secret=6ff108122cce3b0b45e0abf374c14ef4 dtmfmode=rfc2833 canreinvite=no context=from-internal host=dynamic trustrpid=yes sendrpid=no type=friend nat=yes port=5060 qualify=yes qualifyfreq=60 transport=udp avpf=no icesupport=no dtlsenable=no dtlsverify=no dtlssetup=actpass encryption=no callgroup= pickupgroup= dial=SIP/1000 mailbox=1000@device permit=0.0.0.0/0.0.0.0 callerid=Usuario 1 elx4 <1000> callcounter=yes faxdetect=no [1001] deny=0.0.0.0/0.0.0.0 secret=ce93963b0751ed9a88ec1badbc073fce dtmfmode=rfc2833 canreinvite=no context=from-internal host=dynamic trustrpid=yes sendrpid=no type=friend nat=yes port=5060 qualify=yes qualifyfreq=60 transport=wss,ws,udp,tcp,tls avpf=yes icesupport=yes dtlsenable=yes dtlsverify=no dtlssetup=actpass dtlscertfile=/var/lib/asterisk/keys/localhost.crt dtlsprivatekey=/var/lib/asterisk/keys/localhost.key encryption=yes callgroup= pickupgroup= dial=SIP/1001 mailbox=1001@device permit=0.0.0.0/0.0.0.0 callerid=Usuario Alex <1001> callcounter=yes faxdetect=no With this setup, I can make calls using the SIP softphone as usual, both into and out of asterisk. After approving the certificate exceptions, I can also use the webrtc account to generate a call from either Firefox or Google Chrome into asterisk, out to the SIP softphone. The problem arises when I try to make asterisk send a call into the browser. When using Firefox 43 I can receive the call normally (this required patching around ASTERISK-25659) and all is well. However, in Google Chrome, the call is rejected with a message of "Failed to set remote video description send parameters.." as shown in this SIP trace in the browser console: Wed Jan 20 2016 13:54:53 GMT-0500 (ECT) | sip.transport | received WebSocket text message: INVITE sip:8dgpkoa2@192.0.2.210;transport=wss SIP/2.0 Via: SIP/2.0/WS 10.1.0.4:5060;branch=z9hG4bK4f80b96d;rport Max-Forwards: 70 From: "Anonymous" ;tag=as37a33245 To: Contact: Call-ID: 61c6be5b44d587c80b56f98e756037ab@10.1.0.4:5060 CSeq: 102 INVITE User-Agent: FPBX-2.11.0(11.20.0) Date: Wed, 20 Jan 2016 18:54:52 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 1799 v=0 o=root 469858785 469858785 IN IP4 10.1.0.4 s=Asterisk PBX 11.21.0 c=IN IP4 10.1.0.4 b=CT:384 t=0 0 m=audio 14814 UDP/TLS/RTP/SAVPF 4 0 3 8 18 110 9 101 a=rtpmap:4 G723/8000 a=fmtp:4 annexa=no a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:110 speex/8000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=ice-ufrag:0e746cd50c88ce6e383ff3882acebb80 a=ice-pwd:1a9a09862254ae253f06a0bb184fd1b5 a=candidate:Ha010004 1 UDP 2130706431 10.1.0.4 14814 typ host a=candidate:Hc0a80592 1 UDP 2130706431 192.168.5.146 14814 typ host a=candidate:Ha010004 2 UDP 2130706430 10.1.0.4 14815 typ host a=candidate:Hc0a80592 2 UDP 2130706430 192.168.5.146 14815 typ host a=connection:new a=setup:actpass a=fingerprint:SHA-256 A1:9E:D0:11:26:AA:30:00:AF:06:87:9C:A7:C2:70:4F:A3:3F:89:B5:7D:5C:FA:90:89:7E:D0:8A:F0:72:F5:2A a=sendrecv m=v
[asterisk-users] Incoming webrtc call succeeds in Firefox but fails in Google Chrome
I am having trouble getting Google Chrome to accept a WebRTC call coming from Asterisk, even though Firefox can (now) accept the same call without issue. My setup is as follows: Server: CentOS 7 x86_64 (Elastix 4 RC) with IP: 10.1.0.4 192.168.5.146 asterisk-11.21.0 patched to work around https://issues.asterisk.org/jira/browse/ASTERISK-25659 openssl-1.0.1e-51.el7_2.2.x86_64 [root@elx4 ~]# openssl version OpenSSL 1.0.1e-fips 11 Feb 2013 [root@elx4 ~]# openssl ecparam -list_curves secp384r1 : NIST/SECG curve over a 384 bit prime field secp521r1 : NIST/SECG curve over a 521 bit prime field prime256v1: X9.62/SECG curve over a 256 bit prime field Client: Fedora 23 x86_64 Linphone (linphone-3.6.1-10.fc23.x86_64) Firefox 43 (firefox-43.0.3-1.fc23.x86_64) Google Chrome (google-chrome-stable-47.0.2526.111-1.x86_64) SIP.js 0.7.2 I set up my SIP configuration to have two SIP accounts. Account 1000 is the Linphone and 1001 is the webrtc: [general] faxdetect=no vmexten=*97 context=from-sip-external callerid=Unknown notifyringing=yes notifyhold=yes tos_sip=cs3 tos_audio=ef tos_video=af41 alwaysauthreject=yes useragent=FPBX-2.11.0(11.20.0) disallow=all allow=g723 allow=ulaw allow=gsm allow=alaw allow=g729 allow=speex allow=g722 allow=h264 allow=h263p allow=h263 allow=h261 tlsenable=yes tlsbindaddr=0.0.0.0 tlscipher=ALL tlsclientmethod=tlsv1 tlscertfile=/etc/asterisk/keys/asterisk.pem tlscafile=/etc/asterisk/keys/ca.crt callevents=no jbenable=no videosupport=yes allowguest=no srvlookup=no defaultexpiry=120 minexpiry=60 maxexpiry=3600 registerattempts=0 registertimeout=20 g726nonstandard=no maxcallbitrate=384 canreinvite=no rtptimeout=30 rtpholdtimeout=300 rtpkeepalive=0 checkmwi=10 notifyringing=yes notifyhold=yes nat=yes [1000] deny=0.0.0.0/0.0.0.0 secret=6ff108122cce3b0b45e0abf374c14ef4 dtmfmode=rfc2833 canreinvite=no context=from-internal host=dynamic trustrpid=yes sendrpid=no type=friend nat=yes port=5060 qualify=yes qualifyfreq=60 transport=udp avpf=no icesupport=no dtlsenable=no dtlsverify=no dtlssetup=actpass encryption=no callgroup= pickupgroup= dial=SIP/1000 mailbox=1000@device permit=0.0.0.0/0.0.0.0 callerid=Usuario 1 elx4 <1000> callcounter=yes faxdetect=no [1001] deny=0.0.0.0/0.0.0.0 secret=ce93963b0751ed9a88ec1badbc073fce dtmfmode=rfc2833 canreinvite=no context=from-internal host=dynamic trustrpid=yes sendrpid=no type=friend nat=yes port=5060 qualify=yes qualifyfreq=60 transport=wss,ws,udp,tcp,tls avpf=yes icesupport=yes dtlsenable=yes dtlsverify=no dtlssetup=actpass dtlscertfile=/var/lib/asterisk/keys/localhost.crt dtlsprivatekey=/var/lib/asterisk/keys/localhost.key encryption=yes callgroup= pickupgroup= dial=SIP/1001 mailbox=1001@device permit=0.0.0.0/0.0.0.0 callerid=Usuario Alex <1001> callcounter=yes faxdetect=no With this setup, I can make calls using the SIP softphone as usual, both into and out of asterisk. After approving the certificate exceptions, I can also use the webrtc account to generate a call from either Firefox or Google Chrome into asterisk, out to the SIP softphone. The problem arises when I try to make asterisk send a call into the browser. When using Firefox 43 I can receive the call normally (this required patching around ASTERISK-25659) and all is well. However, in Google Chrome, the call is rejected with a message of "Failed to set remote video description send parameters.." as shown in this SIP trace in the browser console: Wed Jan 20 2016 13:54:53 GMT-0500 (ECT) | sip.transport | received WebSocket text message: INVITE sip:8dgpkoa2@192.0.2.210;transport=wss SIP/2.0 Via: SIP/2.0/WS 10.1.0.4:5060;branch=z9hG4bK4f80b96d;rport Max-Forwards: 70 From: "Anonymous" ;tag=as37a33245 To: Contact: Call-ID: 61c6be5b44d587c80b56f98e756037ab@10.1.0.4:5060 CSeq: 102 INVITE User-Agent: FPBX-2.11.0(11.20.0) Date: Wed, 20 Jan 2016 18:54:52 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 1799 v=0 o=root 469858785 469858785 IN IP4 10.1.0.4 s=Asterisk PBX 11.21.0 c=IN IP4 10.1.0.4 b=CT:384 t=0 0 m=audio 14814 UDP/TLS/RTP/SAVPF 4 0 3 8 18 110 9 101 a=rtpmap:4 G723/8000 a=fmtp:4 annexa=no a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:110 speex/8000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=ice-ufrag:0e746cd50c88ce6e383ff3882acebb80 a=ice-pwd:1a9a09862254ae253f06a0bb184fd1b5 a=candidate:Ha010004 1 UDP 2130706431 10.1.0.4 14814 typ host a=candidate:Hc0a80592 1 UDP 2130706431 192.168.5.146 14814 typ host a=candidate:Ha010004 2 UDP 2130706430 10.1.0.4 14815 typ host a=candidate:Hc0a80592 2 UDP 2130706430 192.168.5.146 14815 typ host a=connection:new a=setup:actpass a=fingerprint:SHA-256 A1:9E:D0:11:26:AA:30:00:AF:06:87:9C:A7:C2:70:4F:A3:3F:89:B5:7D:5C:FA:90:89:7E:D0:8A:F0:72:F5:2A a=sendrecv m=video 13042 UDP/TLS/RTP/SAVPF 99 98 34 31 a=ice-ufrag:68da
Re: [asterisk-users] Incoming calls get 488 error
On 8/21/15 6:45 PM, Technical Support wrote: I got a new SNOM M65 which works fine for outgoing calls, but incoming calls never ring at the handset. I captured the SIP traffic and see that my M65 is replying with an "488 not acceptable here". From what I read this is usually codec related but both asterisk and the M65 are set for ulaw as first choice. Looks like the SNOM does not accept the video call. Maybe you should look into why the Asterisk is trying to use video in the first place. I have a SIP trace below. Can someone suggest why the 488 is being generated? --- Received from udp:192.168.253.4:5060 at 21/08/2015 18:37:00 (1198 bytes) INVITE sip:290006@192.168.253.20;line=14994 SIP/2.0 Via: SIP/2.0/UDP 192.168.253.4:5060;branch=z9hG4bK4990095a Max-Forwards: 70 From: "test user" ;tag=as7b616c8d To: Contact: Call-ID: 36334383058109cd2325341a0f18ac79@192.168.253.4:5060 CSeq: 102 INVITE User-Agent: FPBX-2.11.0(11.10.2) Date: Fri, 21 Aug 2015 22:37:02 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 606 v=0 o=root 1678280845 1678280845 IN IP4 192.168.253.4 s=Asterisk PBX 11.10.2 c=IN IP4 192.168.253.4 b=CT:384 t=0 0 m=audio 18090 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv m=video 12226 RTP/AVP 99 98 34 31 a=rtpmap:99 H264/9 a=fmtp:99 redundant-pic-cap=0;parameter-add=0;packetization-mode=0;level-asymmetry-allowed=0 a=rtpmap:98 H263-1998/9 a=fmtp:98 F=0;I=0;J=0;T=0;K=0;N=0;BPP=0;HRD=0 a=rtpmap:34 H263/9 a=fmtp:34 F=0;I=0;J=0;T=0;K=0;N=0;BPP=0;HRD=0 a=rtpmap:31 H261/9 a=sendrecv Sent to udp:192.168.253.4:5060 at 21/08/2015 18:37:00 (280 bytes) SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.253.4:5060;branch=z9hG4bK4990095a From: "test user" ;tag=as7b616c8d To: Call-ID: 36334383058109cd2325341a0f18ac79@192.168.253.4:5060 CSeq: 102 INVITE Content-Length: 0 Sent to udp:192.168.253.4:5060 at 21/08/2015 18:37:00 (441 bytes) SIP/2.0 488 Not Acceptable Here Via: SIP/2.0/UDP 192.168.253.4:5060;branch=z9hG4bK4990095a Max-Forwards: 70 From: "test user" ;tag=as7b616c8d To: ;tag=ld65q Call-ID: 36334383058109cd2325341a0f18ac79@192.168.253.4:5060 CSeq: 102 INVITE Contact: User-Agent: snomM700/03.24.0007 (MAC=0004136103FB; SER= 0; HW=255) Content-Length: 0 -- Technical Support http://www.cellroute.net -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming calls get 488 error
Hi, By the sip trace is very difficult to tell because the SIP messages are fine. Try to enable all codec, and if possible copy and paste your asterisk sip configuration for this peer. Enviado do meu telefone Android usando o Symantec TouchDown (www.symantec.com) -Original Message- From: Technical Support [supp...@telium.ca] Received: sexta-feira, 21 ago 2015, 19:46 To: asterisk-users@lists.digium.com [asterisk-users@lists.digium.com] Subject: [asterisk-users] Incoming calls get 488 error I got a new SNOM M65 which works fine for outgoing calls, but incoming calls never ring at the handset. I captured the SIP traffic and see that my M65 is replying with an "488 not acceptable here". From what I read this is usually codec related but both asterisk and the M65 are set for ulaw as first choice. I have a SIP trace below. Can someone suggest why the 488 is being generated? --- Received from udp:192.168.253.4:5060 at 21/08/2015 18:37:00 (1198 bytes) INVITE sip:290006@192.168.253.20;line=14994 SIP/2.0 Via: SIP/2.0/UDP 192.168.253.4:5060;branch=z9hG4bK4990095a Max-Forwards: 70 From: "test user" ;tag=as7b616c8d To: Contact: Call-ID: 36334383058109cd2325341a0f18ac79@192.168.253.4:5060 CSeq: 102 INVITE User-Agent: FPBX-2.11.0(11.10.2) Date: Fri, 21 Aug 2015 22:37:02 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 606 v=0 o=root 1678280845 1678280845 IN IP4 192.168.253.4 s=Asterisk PBX 11.10.2 c=IN IP4 192.168.253.4 b=CT:384 t=0 0 m=audio 18090 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv m=video 12226 RTP/AVP 99 98 34 31 a=rtpmap:99 H264/9 a=fmtp:99 redundant-pic-cap=0;parameter-add=0;packetization-mode=0;level-asymmetry-allowed=0 a=rtpmap:98 H263-1998/9 a=fmtp:98 F=0;I=0;J=0;T=0;K=0;N=0;BPP=0;HRD=0 a=rtpmap:34 H263/9 a=fmtp:34 F=0;I=0;J=0;T=0;K=0;N=0;BPP=0;HRD=0 a=rtpmap:31 H261/9 a=sendrecv Sent to udp:192.168.253.4:5060 at 21/08/2015 18:37:00 (280 bytes) SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.253.4:5060;branch=z9hG4bK4990095a From: "test user" ;tag=as7b616c8d To: Call-ID: 36334383058109cd2325341a0f18ac79@192.168.253.4:5060 CSeq: 102 INVITE Content-Length: 0 Sent to udp:192.168.253.4:5060 at 21/08/2015 18:37:00 (441 bytes) SIP/2.0 488 Not Acceptable Here Via: SIP/2.0/UDP 192.168.253.4:5060;branch=z9hG4bK4990095a Max-Forwards: 70 From: "test user" ;tag=as7b616c8d To: ;tag=ld65q Call-ID: 36334383058109cd2325341a0f18ac79@192.168.253.4:5060 CSeq: 102 INVITE Contact: User-Agent: snomM700/03.24.0007 (MAC=0004136103FB; SER= 0; HW=255) Content-Length: 0 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Incoming calls get 488 error
I got a new SNOM M65 which works fine for outgoing calls, but incoming calls never ring at the handset. I captured the SIP traffic and see that my M65 is replying with an "488 not acceptable here". From what I read this is usually codec related but both asterisk and the M65 are set for ulaw as first choice. I have a SIP trace below. Can someone suggest why the 488 is being generated? --- Received from udp:192.168.253.4:5060 at 21/08/2015 18:37:00 (1198 bytes) INVITE sip:290006@192.168.253.20;line=14994 SIP/2.0 Via: SIP/2.0/UDP 192.168.253.4:5060;branch=z9hG4bK4990095a Max-Forwards: 70 From: "test user" ;tag=as7b616c8d To: Contact: Call-ID: 36334383058109cd2325341a0f18ac79@192.168.253.4:5060 CSeq: 102 INVITE User-Agent: FPBX-2.11.0(11.10.2) Date: Fri, 21 Aug 2015 22:37:02 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 606 v=0 o=root 1678280845 1678280845 IN IP4 192.168.253.4 s=Asterisk PBX 11.10.2 c=IN IP4 192.168.253.4 b=CT:384 t=0 0 m=audio 18090 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv m=video 12226 RTP/AVP 99 98 34 31 a=rtpmap:99 H264/9 a=fmtp:99 redundant-pic-cap=0;parameter-add=0;packetization-mode=0;level-asymmetry-allowed=0 a=rtpmap:98 H263-1998/9 a=fmtp:98 F=0;I=0;J=0;T=0;K=0;N=0;BPP=0;HRD=0 a=rtpmap:34 H263/9 a=fmtp:34 F=0;I=0;J=0;T=0;K=0;N=0;BPP=0;HRD=0 a=rtpmap:31 H261/9 a=sendrecv Sent to udp:192.168.253.4:5060 at 21/08/2015 18:37:00 (280 bytes) SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.253.4:5060;branch=z9hG4bK4990095a From: "test user" ;tag=as7b616c8d To: Call-ID: 36334383058109cd2325341a0f18ac79@192.168.253.4:5060 CSeq: 102 INVITE Content-Length: 0 Sent to udp:192.168.253.4:5060 at 21/08/2015 18:37:00 (441 bytes) SIP/2.0 488 Not Acceptable Here Via: SIP/2.0/UDP 192.168.253.4:5060;branch=z9hG4bK4990095a Max-Forwards: 70 From: "test user" ;tag=as7b616c8d To: ;tag=ld65q Call-ID: 36334383058109cd2325341a0f18ac79@192.168.253.4:5060 CSeq: 102 INVITE Contact: User-Agent: snomM700/03.24.0007 (MAC=0004136103FB; SER= 0; HW=255) Content-Length: 0 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Incoming calls to a GSM gateway & "SIP/2.0 401 Unauthorized" response when dial 7777 to Asterisk
Hello: I'm newbie in asterisk, please help me. My context is as follows: 192.168.4.2 --> Asterisk 11.13.1 complied from source 192.168.4.4 --> Yeastar NeoGate TG100 GSM gateway When I call from a GSM cell phone, my TG100 GSM gateway answers and dials extension (configured as a hotline on TG100) to asterisk server, but asterisk server sends me "SIP/2.0 401 Unauthorized" response, I think it's a matter of contexts but I don't find the problem. Attached are sip.conf, extensions.conf and debug from 192.168.4.4 (TG100 GSM gateway). Thanks in advance. <--- SIP read from UDP:192.168.4.4:5060 ---> INVITE sip:@192.168.4.2:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.4.4:5060;branch=z9hG4bK0b7c4652;rport Max-Forwards: 70 From: "9" ;tag=as67354416 To: Contact: Call-ID: 12663beb04ae514f10c4b3a145368d5c@192.168.4.4 CSeq: 102 INVITE User-Agent: TG100 Date: Wed, 12 Nov 2014 10:13:38 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 281 v=0 o=root 1426707418 1426707418 IN IP4 192.168.4.4 s=Asterisk PBX 1.6.2.6 c=IN IP4 192.168.4.4 t=0 0 m=audio 10048 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <-> --- (14 headers 13 lines) --- Sending to 192.168.4.4:5060 (no NAT) Sending to 192.168.4.4:5060 (no NAT) Using INVITE request as basis request - 12663beb04ae514f10c4b3a145368d5c@192.168.4.4 Found peer '5' for '9' from 192.168.4.4:5060 <--- Reliably Transmitting (no NAT) to 192.168.4.4:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.4.4:5060;branch=z9hG4bK0b7c4652;received=192.168.4.4;rport=5060 From: "9" ;tag=as67354416 To: ;tag=as16de6e5c Call-ID: 12663beb04ae514f10c4b3a145368d5c@192.168.4.4 CSeq: 102 INVITE Server: Asterisk PBX 11.13.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="72011a6b" Content-Length: 0 <> Scheduling destruction of SIP dialog '12663beb04ae514f10c4b3a145368d5c@192.168.4.4' in 6400 ms (Method: INVITE) <--- SIP read from UDP:192.168.4.4:5060 ---> ACK sip:@192.168.4.2:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.4.4:5060;branch=z9hG4bK0b7c4652;rport Max-Forwards: 70 From: "9" ;tag=as67354416 To: ;tag=as16de6e5c Contact: Call-ID: 12663beb04ae514f10c4b3a145368d5c@192.168.4.4 CSeq: 102 ACK User-Agent: TG100 Content-Length: 0 <-> --- (10 headers 0 lines) --- Really destroying SIP dialog '6e9eab843b74d0860b108ed13a5d22c9@192.168.4.4' Method: OPTIONS Really destroying SIP dialog '12663beb04ae514f10c4b3a145368d5c@192.168.4.4' Method: ACK uc*CLI> extensions.conf Description: Binary data sip.conf Description: Binary data -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] incoming calls fall into echo test mode
On Saturday 19 Jul 2014, Norman Molhant wrote: > I tried many things on our FreePBX box and found out > the problem seems somehow linked with the customer's > extension (or phone number), not his inbound route > (changing the latter has no effect on the problem). > > Creating a new extension with another phone number > would solve the problem (I tried it and it works), > but this customer wants to keep his current phone > number and when I tried deleting his extension then > creating a new one with his current phone number, > the new extension presented the same problem as the > previous one... > > Anyone knows what could cause such a problem and/or > how to solve it ? You really have supplied incomplete information here, by neglecting to mention the actual extension number which is causing the problems. That would have had somebody onto it like a shot. What follows is an educated guess based on the most likely scenario according to the available information: Somewhere in your dialplan, probably in a section that has already been "helpfully" configured for you by FreePBX, the extension number you assigned to your customer has been appropriated for an echotest. I suggest to grep for (firstly) the extension number in question, and (if that does not work, perhaps because the echotest is a wildcard match aot a literal one) then search instead for 'exten[ ]*=>' (afraid that one will give you many more hits . you'll have to look through them yourself) under /etc/asterisk. Use the -R option to search subfolders as well. -- AJS Note: Originating address only accepts e-mail from list! If replying off- list, change address to asterisk1list at earthshod dot co dot uk . -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] incoming calls fall into echo test mode
check your logs /var/log/asterisk/full -- make sure your verbosity is set high enough to do you good and you wll probably find the answer. Pat Collins wrote: > Perhaps assigned as a test number somewhere along the line? > Are these ISDN, SIP, IAX calls? > There are MANY smart people on this list. > Maybe sharing the relevant configs and traces is a good place to start??? > > -Original Message- > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Norman Molhant > Sent: Saturday, July 19, 2014 10:43 AM > To: asterisk-users@lists.digium.com > Subject: [asterisk-users] incoming calls fall into echo test mode > > Hello all, > > Weird trouble here: > we have 60-some happy subscribers on a FreePBX box, each with its own phone > number, with no problem at all, except for one (and only one) subscriber who > has this > problem: his outgoing calls are ok, but when someone dials his phone number > (be it from our network or from any other place in the world), the caller > ears the standard message signalling he has entered the echo test mode and > must dial # to exit that mode. > > Most callers don't understand what's going on, then give up and hang up > without dialling #. Very few dial # one or more times, then those few get > our customer's phone ringing and are then able to reach our customer. > > I went through all the docs, wikis and discussions I found on the web, > without finding any data on how to solve that problem. > > I tried many things on our FreePBX box and found out the problem seems > somehow linked with the customer's extension (or phone number), not his > inbound route (changing the latter has no effect on the problem). > > Creating a new extension with another phone number would solve the problem > (I tried it and it works), but this customer wants to keep his current phone > number and when I tried deleting his extension then creating a new one with > his current phone number, the new extension presented the same problem as > the previous one... > > Anyone knows what could cause such a problem and/or how to solve it ? > > Thanks, > Norman. > ad...@csur.ca > > > > > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to > Asterisk? Join us for a live introductory webinar every Thurs: >http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: >http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] incoming calls fall into echo test mode
Perhaps assigned as a test number somewhere along the line? Are these ISDN, SIP, IAX calls? There are MANY smart people on this list. Maybe sharing the relevant configs and traces is a good place to start??? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Norman Molhant Sent: Saturday, July 19, 2014 10:43 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] incoming calls fall into echo test mode Hello all, Weird trouble here: we have 60-some happy subscribers on a FreePBX box, each with its own phone number, with no problem at all, except for one (and only one) subscriber who has this problem: his outgoing calls are ok, but when someone dials his phone number (be it from our network or from any other place in the world), the caller ears the standard message signalling he has entered the echo test mode and must dial # to exit that mode. Most callers don't understand what's going on, then give up and hang up without dialling #. Very few dial # one or more times, then those few get our customer's phone ringing and are then able to reach our customer. I went through all the docs, wikis and discussions I found on the web, without finding any data on how to solve that problem. I tried many things on our FreePBX box and found out the problem seems somehow linked with the customer's extension (or phone number), not his inbound route (changing the latter has no effect on the problem). Creating a new extension with another phone number would solve the problem (I tried it and it works), but this customer wants to keep his current phone number and when I tried deleting his extension then creating a new one with his current phone number, the new extension presented the same problem as the previous one... Anyone knows what could cause such a problem and/or how to solve it ? Thanks, Norman. ad...@csur.ca -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] incoming calls fall into echo test mode
You might get a better response on the FreePBX forum. (FreePBX adds pre-built dialplan elements onto standard asterisk. This forum is more for Asterisk) But some suggestions: SSH to your PBX enter the Asterisk CLI set verbose to 10 Call into the problematic number ...and watch where the call is being misrouted in the dialplan From: asterisk-users-boun...@lists.digium.com on behalf of Norman Molhant Sent: Saturday, July 19, 2014 10:43 AM To: Asterisk Users List Subject: [asterisk-users] incoming calls fall into echo test mode Hello all, Weird trouble here: we have 60-some happy subscribers on a FreePBX box, each with its own phone number, with no problem at all, except for one (and only one) subscriber who has this problem: his outgoing calls are ok, but when someone dials his phone number (be it from our network or from any other place in the world), the caller ears the standard message signalling he has entered the echo test mode and must dial # to exit that mode. Most callers don't understand what's going on, then give up and hang up without dialling #. Very few dial # one or more times, then those few get our customer's phone ringing and are then able to reach our customer. I went through all the docs, wikis and discussions I found on the web, without finding any data on how to solve that problem. I tried many things on our FreePBX box and found out the problem seems somehow linked with the customer's extension (or phone number), not his inbound route (changing the latter has no effect on the problem). Creating a new extension with another phone number would solve the problem (I tried it and it works), but this customer wants to keep his current phone number and when I tried deleting his extension then creating a new one with his current phone number, the new extension presented the same problem as the previous one... Anyone knows what could cause such a problem and/or how to solve it ? Thanks, Norman. ad...@csur.ca -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] incoming calls fall into echo test mode
Hello all, Weird trouble here: we have 60-some happy subscribers on a FreePBX box, each with its own phone number, with no problem at all, except for one (and only one) subscriber who has this problem: his outgoing calls are ok, but when someone dials his phone number (be it from our network or from any other place in the world), the caller ears the standard message signalling he has entered the echo test mode and must dial # to exit that mode. Most callers don't understand what's going on, then give up and hang up without dialling #. Very few dial # one or more times, then those few get our customer's phone ringing and are then able to reach our customer. I went through all the docs, wikis and discussions I found on the web, without finding any data on how to solve that problem. I tried many things on our FreePBX box and found out the problem seems somehow linked with the customer's extension (or phone number), not his inbound route (changing the latter has no effect on the problem). Creating a new extension with another phone number would solve the problem (I tried it and it works), but this customer wants to keep his current phone number and when I tried deleting his extension then creating a new one with his current phone number, the new extension presented the same problem as the previous one... Anyone knows what could cause such a problem and/or how to solve it ? Thanks, Norman. ad...@csur.ca -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming Fax Issue with Asterisk 11.7 and Digium Fax
On 3/02/2014 8:42 PM, Jakob-Matthias Böttger wrote: Am 03.02.2014 13:20, schrieb Jakob-Matthias Böttger: Am 03.02.2014 12:56, schrieb Larry Moore: On 3/02/2014 7:38 PM, Jakob-Matthias Böttger wrote: as He is describing it he should actually provide t.38. but i don't know why it is not working thus im now getting Feb 3 12:32:55] WARNING[9942][C-0004]: chan_sip.c:10353 process_sdp: Failed to initialize UDPTL, declining image stream [Feb 3 12:32:55] WARNING[9942][C-0004]: chan_sip.c:10497 process_sdp: Insufficient information in SDP (c=)... and then the fax session starts recording data In udptl.conf set use_even_ports=yes and then issue a reload. You can confirm the settings have been applied by performing udptl show config. Change the the t38 line to read as; t38pt_udptl=yes,redundancy,maxdatagram=400 Reload sip and test. after that i started udptl debug as well and now i'm getting lots of UDPTL (SIP/sipcall.ch-0007): packet to 212.117.203.76:24492 (seq 152, len 11) and in between [Feb 3 13:18:30] WARNING[26313][C-0007]: res_rtp_asterisk.c:3548 ast_rtp_read: RTP Read too short and in the end [Feb 3 13:18:37] WARNING[9942]: chan_sip.c:4409 __sip_autodestruct: Autodestruct on dialog '24d15e0d-28df847a-9fae13c-7...@sip.iforb.com~1o' with owner SIP/sipcall.ch-0007 in place (Method: BYE). Rescheduling destruction for 1 ms [Feb 3 13:18:41] ERROR[26313][C-0007]: res_fax.c:1535 generic_fax_exec: channel 'SIP/sipcall.ch-0007' FAX session '7' failure, reason: 'fax session timed-out' (TIMEOUT) == Spawn extension (fax-rx, receive, 11) exited non-zero on 'SIP/sipcall.ch-0007' Thx, Jakob may do i have to open more ports then udp 1:2 (RTP), udp 4000:4999 (UDPTL) and tcp 5060,5061(SIP/TLS) The T.38 connection will be attempted when ReceiveFax is called. The port number to use should be in the SDP information, yes, allow udp ports 4000-4999 in and out. If your firewall can be so configured you could set it to allow traffic in and out based upon the user ID Asterisk is running as, assuming it is using a unique unprivileged id. You may like to try the following to see if your SIP provider will initiate a T.38 re-invite. sip.conf [general] faxdetect=t38 [sipcall.ch] directmedia=no In extensions.conf change Wait(2) to Wait(5), if your VSP sends you a T.38 re-invite this will trigger the switch to the Fax extension. If this proves successful you can work on removing the Wait() from your dialplan as Asterisk will remain in the audio path and should successfully switch to the fax extension if extension 200 or 201 answer a call that happens to be a fax. Larry. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming Fax Issue with Asterisk 11.7 and Digium Fax
Am 03.02.2014 13:20, schrieb Jakob-Matthias Böttger: Am 03.02.2014 12:56, schrieb Larry Moore: On 3/02/2014 7:38 PM, Jakob-Matthias Böttger wrote: as He is describing it he should actually provide t.38. but i don't know why it is not working thus im now getting Feb 3 12:32:55] WARNING[9942][C-0004]: chan_sip.c:10353 process_sdp: Failed to initialize UDPTL, declining image stream [Feb 3 12:32:55] WARNING[9942][C-0004]: chan_sip.c:10497 process_sdp: Insufficient information in SDP (c=)... and then the fax session starts recording data In udptl.conf set use_even_ports=yes and then issue a reload. You can confirm the settings have been applied by performing udptl show config. Change the the t38 line to read as; t38pt_udptl=yes,redundancy,maxdatagram=400 Reload sip and test. after that i started udptl debug as well and now i'm getting lots of UDPTL (SIP/sipcall.ch-0007): packet to 212.117.203.76:24492 (seq 152, len 11) and in between [Feb 3 13:18:30] WARNING[26313][C-0007]: res_rtp_asterisk.c:3548 ast_rtp_read: RTP Read too short and in the end [Feb 3 13:18:37] WARNING[9942]: chan_sip.c:4409 __sip_autodestruct: Autodestruct on dialog '24d15e0d-28df847a-9fae13c-7...@sip.iforb.com~1o' with owner SIP/sipcall.ch-0007 in place (Method: BYE). Rescheduling destruction for 1 ms [Feb 3 13:18:41] ERROR[26313][C-0007]: res_fax.c:1535 generic_fax_exec: channel 'SIP/sipcall.ch-0007' FAX session '7' failure, reason: 'fax session timed-out' (TIMEOUT) == Spawn extension (fax-rx, receive, 11) exited non-zero on 'SIP/sipcall.ch-0007' Thx, Jakob may do i have to open more ports then udp 1:2 (RTP), udp 4000:4999 (UDPTL) and tcp 5060,5061(SIP/TLS) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming Fax Issue with Asterisk 11.7 and Digium Fax
Am 03.02.2014 12:56, schrieb Larry Moore: On 3/02/2014 7:38 PM, Jakob-Matthias Böttger wrote: as He is describing it he should actually provide t.38. but i don't know why it is not working thus im now getting Feb 3 12:32:55] WARNING[9942][C-0004]: chan_sip.c:10353 process_sdp: Failed to initialize UDPTL, declining image stream [Feb 3 12:32:55] WARNING[9942][C-0004]: chan_sip.c:10497 process_sdp: Insufficient information in SDP (c=)... and then the fax session starts recording data In udptl.conf set use_even_ports=yes and then issue a reload. You can confirm the settings have been applied by performing udptl show config. Change the the t38 line to read as; t38pt_udptl=yes,redundancy,maxdatagram=400 Reload sip and test. after that i started udptl debug as well and now i'm getting lots of UDPTL (SIP/sipcall.ch-0007): packet to 212.117.203.76:24492 (seq 152, len 11) and in between [Feb 3 13:18:30] WARNING[26313][C-0007]: res_rtp_asterisk.c:3548 ast_rtp_read: RTP Read too short and in the end [Feb 3 13:18:37] WARNING[9942]: chan_sip.c:4409 __sip_autodestruct: Autodestruct on dialog '24d15e0d-28df847a-9fae13c-7...@sip.iforb.com~1o' with owner SIP/sipcall.ch-0007 in place (Method: BYE). Rescheduling destruction for 1 ms [Feb 3 13:18:41] ERROR[26313][C-0007]: res_fax.c:1535 generic_fax_exec: channel 'SIP/sipcall.ch-0007' FAX session '7' failure, reason: 'fax session timed-out' (TIMEOUT) == Spawn extension (fax-rx, receive, 11) exited non-zero on 'SIP/sipcall.ch-0007' Thx, Jakob -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming Fax Issue with Asterisk 11.7 and Digium Fax
On 3/02/2014 7:38 PM, Jakob-Matthias Böttger wrote: as He is describing it he should actually provide t.38. but i don't know why it is not working thus im now getting Feb 3 12:32:55] WARNING[9942][C-0004]: chan_sip.c:10353 process_sdp: Failed to initialize UDPTL, declining image stream [Feb 3 12:32:55] WARNING[9942][C-0004]: chan_sip.c:10497 process_sdp: Insufficient information in SDP (c=)... and then the fax session starts recording data In udptl.conf set use_even_ports=yes and then issue a reload. You can confirm the settings have been applied by performing udptl show config. Change the the t38 line to read as; t38pt_udptl=yes,redundancy,maxdatagram=400 Reload sip and test. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming Fax Issue with Asterisk 11.7 and Digium Fax
as He is describing it he should actually provide t.38. but i don't know why it is not working thus im now getting Feb 3 12:32:55] WARNING[9942][C-0004]: chan_sip.c:10353 process_sdp: Failed to initialize UDPTL, declining image stream [Feb 3 12:32:55] WARNING[9942][C-0004]: chan_sip.c:10497 process_sdp: Insufficient information in SDP (c=)... and then the fax session starts recording data Am 03.02.2014 12:34, schrieb Larry Moore: On 3/02/2014 7:15 PM, Jakob-Matthias Böttger wrote: Hi, changing faxdetect=cng and t38pt_udptl=no helped making it work. Hmm, the fax will be received as an audio call rather than T.38, setting t38pt_udptl=no has turned off T.38. Do you know if your upstream provider supports T.38? Larry. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming Fax Issue with Asterisk 11.7 and Digium Fax
On 3/02/2014 7:15 PM, Jakob-Matthias Böttger wrote: Hi, changing faxdetect=cng and t38pt_udptl=no helped making it work. Hmm, the fax will be received as an audio call rather than T.38, setting t38pt_udptl=no has turned off T.38. Do you know if your upstream provider supports T.38? Larry. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming Fax Issue with Asterisk 11.7 and Digium Fax
Hi, changing faxdetect=cng and t38pt_udptl=no helped making it work. Thanks Am 03.02.2014 11:57, schrieb Larry Moore: On 3/02/2014 3:34 PM, Jakob-Matthias Böttger wrote: . . . [sipcall.ch] type=peer insecure=invite defaultuser=123456789 fromuser=123456789 fromdomain=voipdomain.com secret=gueswhat host=voipdomain.com qualify=yes context=from-sip dtmfmode=rfc2833 callbackextension=123456789 add directmedia=no setvar=FAXOPT(gateway)=no change insecure=port,invite [fax-rx] exten => receive,1,NoOp( FAX RECEIVE ) exten => receive,n,Set(GLOBAL(FAXCOUNT)=$["${GLOBAL(FAXCOUNT)}" + "1"]) exten => receive,n,Set(FAXCOUNT=${GLOBAL(FAXCOUNT)}) Do you want to keep your received faxes or is it OK to overwrite them the next time asterisk is re-started!? udptl.conf [general] udptlstart=4000 udptlend=4999 udptlfecentries = 3 udptlfecspan = 3 use_even_ports = no You may want to change use_even_ports=yes You will need to restart Asterisk for this change. Some other suggestion if the above doesn't help are; faxdetect=cng t38pt_udptl=no Larry. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming Fax Issue with Asterisk 11.7 and Digium Fax
On 3/02/2014 3:34 PM, Jakob-Matthias Böttger wrote: . . . [sipcall.ch] type=peer insecure=invite defaultuser=123456789 fromuser=123456789 fromdomain=voipdomain.com secret=gueswhat host=voipdomain.com qualify=yes context=from-sip dtmfmode=rfc2833 callbackextension=123456789 add directmedia=no setvar=FAXOPT(gateway)=no change insecure=port,invite [fax-rx] exten => receive,1,NoOp( FAX RECEIVE ) exten => receive,n,Set(GLOBAL(FAXCOUNT)=$["${GLOBAL(FAXCOUNT)}" + "1"]) exten => receive,n,Set(FAXCOUNT=${GLOBAL(FAXCOUNT)}) Do you want to keep your received faxes or is it OK to overwrite them the next time asterisk is re-started!? udptl.conf [general] udptlstart=4000 udptlend=4999 udptlfecentries = 3 udptlfecspan = 3 use_even_ports = no You may want to change use_even_ports=yes You will need to restart Asterisk for this change. Some other suggestion if the above doesn't help are; faxdetect=cng t38pt_udptl=no Larry. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Incoming Fax Issue with Asterisk 11.7 and Digium Fax
Hi, im using a Asterisk Server which is not behind NAT. First i had problems with the fax detection. But this is now solved after adding a wait(2) at the correct place. But i'm still unable to receive a fax due to res_rtp_asterisk.c:3548 ast_rtp_read: RTP Read too short after the Fax session has started. My sip.conf includes [general] allowguest=no alwaysauthreject=yes sendrpid=rpid trustrpid=yes language=de videosupport=yes callevents=yes qualify=yes nat=force_rport,comedia faxdetect=yes t38pt_udptl=yes,redundancy,maxdatagram=400 directrtpsetup=yes disallow=all allow=ulaw allow=alaw and the corresponding Peer [sipcall.ch] type=peer insecure=invite defaultuser=123456789 fromuser=123456789 fromdomain=voipdomain.com secret=gueswhat host=voipdomain.com qualify=yes context=from-sip dtmfmode=rfc2833 callbackextension=123456789 the Dialplan [inbound] exten => _X.,1,Answer() exten => _X.,n,Set(DB(lastcaller/number)=${CALLERID(num)}) exten => _X.,n,GotoIf(${BLACKLIST()}?black,1) exten => _X.,n,Wait(2) exten => _X.,n,Dial(SIP/200&SIP/201,60,tToxX) exten => _X.,n,Goto(ausser-zeit,_X.,3) exten => _X.,n,Hangup() exten => fax,1,NoOp( FAX DETECTED ) exten => fax,n,Goto(fax-rx,receive,1) [fax-rx] exten => receive,1,NoOp( FAX RECEIVE ) exten => receive,n,Set(GLOBAL(FAXCOUNT)=$["${GLOBAL(FAXCOUNT)}" + "1"]) exten => receive,n,Set(FAXCOUNT=${GLOBAL(FAXCOUNT)}) exten => receive,n,Set(FAXFILE=fax-${FAXCOUNT}-rx.tif) exten => receive,n,Set(GLOBAL(LASTFAXCALLERoNUM)=${CALLERID(num)}) exten => receive,n,Set(GLOBAL(LASTFAXCALLERNAME)=${CALLERID(name)}) exten => receive,n,NoOp( SETTING FAXOPT ) exten => receive,n,Set(FAXOPT(ecm)=yes) exten => receive,n,Set(FAXOPT(headerinfo)=MYFAX RX) exten => receive,n,Set(FAXOPT(localstationid)=1234567890) exten => receive,n,Set(FAXOPT(maxrate)=14400) exten => receive,n,Set(FAXOPT(minrate)=2400) exten => receive,n,NoOp(FAXOPT(ecm) : ${FAXOPT(ecm)}) exten => receive,n,NoOp(FAXOPT(headerinfo) : ${FAXOPT(headerinfo)}) exten => receive,n,NoOp(FAXOPT(localstationid) : ${FAXOPT(localstationid)}) exten => receive,n,NoOp(FAXOPT(maxrate) : ${FAXOPT(maxrate)}) exten => receive,n,NoOp(FAXOPT(minrate) : ${FAXOPT(minrate)}) exten => receive,n,NoOp( RECEIVING FAX : ${FAXFILE} ) exten => receive,n,ReceiveFAX(/var/spool/asterisk/faxin/${FAXFILE},dfs) exten => h,1,NoOp(FAXOPT(ecm) : ${FAXOPT(ecm)}) exten => h,n,NoOp(FAXOPT(filename) : ${FAXOPT(filename)}) exten => h,n,NoOp(FAXOPT(headerinfo) : ${FAXOPT(headerinfo)}) exten => h,n,NoOp(FAXOPT(localstationid) : ${FAXOPT(localstationid)}) exten => h,n,NoOp(FAXOPT(maxrate) : ${FAXOPT(maxrate)}) exten => h,n,NoOp(FAXOPT(minrate) : ${FAXOPT(minrate)}) exten => h,n,NoOp(FAXOPT(pages) : ${FAXOPT(pages)}) exten => h,n,NoOp(FAXOPT(rate) : ${FAXOPT(rate)}) exten => h,n,NoOp(FAXOPT(remotestationid) : ${FAXOPT(remotestationid)}) exten => h,n,NoOp(FAXOPT(resolution) : ${FAXOPT(resolution)}) exten => h,n,NoOp(FAXOPT(status) : ${FAXOPT(status)}) exten => h,n,NoOp(FAXOPT(statusstr) : ${FAXOPT(statusstr)}) exten => h,n,NoOp(FAXOPT(error) : ${FAXOPT(error)}) udptl.conf [general] udptlstart=4000 udptlend=4999 udptlfecentries = 3 udptlfecspan = 3 use_even_ports = no rtp.conf [general] rtpstart=1 rtpend=2 res_fax.conf [general] maxrate=14400 minrate=2400 statusevents=yes modems=v17,v27,v29 ecm=yes mail*CLI> core set verbose 6 Set remote console verbosity to 6 == Using SIP RTP CoS mark 5 -- Executing [41325122774@from-sip:1] Answer("SIP/sipcall.ch-008d", "") in new stack > 0x7f3964080f30 -- Probation passed - setting RTP source address to 123.456.789.123:20600 Got RTP packet from123.456.789.123:20600 (type 00, seq 042281, ts 1387619622, len 000160) -- Executing [41325122774@from-sip:2] Set("SIP/sipcall.ch-008d", "DB(lastcaller/number)=987654321") in new stack -- Executing [41325122774@from-sip:3] GotoIf("SIP/sipcall.ch-008d", "0?black,1") in new stack -- Executing [41325122774@from-sip:4] Wait("SIP/sipcall.ch-008d", "2") in new stack Got RTP packet from123.456.789.123:20600 (type 00, seq 042282, ts 1387619782, len 000160) Got RTP packet from123.456.789.123:20600 (type 00, seq 042283, ts 1387619942, len 000160) Got RTP packet from123.456.789.123:20600 (type 00, seq 042284, ts 1387620102, len 000160) Got RTP packet from123.456.789.123:20600 (type 00, seq 042285, ts 1387620262, len 000160) Got RTP packet from123.456.789.123:20600 (type 00, seq 042286, ts 1387620422, len 000160) Got RTP packet from123.456.789.123:20600 (type 00, seq 042287, ts 1387620582, len 000160) Got RTP packet from123.456.789.123:20600 (type 00, seq 042288, ts 1387620742, len 000160) Got RTP packet from123.456.789.123:20600 (type 00, seq 042289, ts 1387620902, len 000160) Got RTP packet from123.456.789.123:20600 (type 00, seq 042290, ts 1387621062, len 000160) Got RTP packet from123.456.789.123:20600 (type 00, seq 042291, ts 1387621222, len 000160
Re: [asterisk-users] incoming DAHDI Channel explained
B.H. Hi! On Wed, Jun 5, 2013 at 7:26 PM, jg wrote: For a BRI device a single span has 2 channels, a PRI device up to 30. As > far as channel variables go the actual channel does not seem to get > reported, but this is not really necessary. AFAIK, at least for AMI listeners, the real channel/span is reported by DAHDIChannelEvent attributes: 'dahdichannel' reports the actual DAHDI channel number 'dahdispan' is a span number > > > jg > > -- משיח NOW! Moshiach is coming very soon, prepare yourself! יחי אדוננו מורינו ורבינו מלך המשיח לעולם ועד! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] incoming DAHDI Channel explained
Yes, my assumption was wrong and to make things worse, my CDR data clearly show that "i" cannot denote incoming calls. Maybe it's time that I learn the rules as well: Analog channels do not seem to have a special identifier. The 1st call for analog channel 13 would be s.th. like DAHDI/13-1. Outside calls via an ISDN connection with s.th. like DIAL(DAHDI/r2/08932168,..) would dial the number using DAHDI group 2 in a round robin fashion, but internally the channel would be s.th. like DAHDI/iX/08932168-abcd. The span X is not related to the dial group and depends on the configuration. For a BRI device a single span has 2 channels, a PRI device up to 30. As far as channel variables go the actual channel does not seem to get reported, but this is not really necessary. jg -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] incoming DAHDI Channel explained
> Sangoma's tech support is probably the better source of information. > > DAHDI: obviously DAHDI channel > i: incoming call The 'i' is for ISDN not incoming call since it will be this way for outgoing calls as well. > 3: span 3 (not the port) > 211123456: CLID, probably subject to filtering (see > national/international prefix settings) > 89c: internal counter (i.e. 2204 calls so far) The other fields are pretty much as described by jg. Richard -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] incoming DAHDI Channel explained
Sangoma's tech support is probably the better source of information. DAHDI: obviously DAHDI channel i: incoming call 3: span 3 (not the port) 211123456: CLID, probably subject to filtering (see national/international prefix settings) 89c: internal counter (i.e. 2204 calls so far) jg -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] incoming DAHDI Channel explained
Hi, I use a Sangoma A104d-Card (with 4 x germany E1). I process some calls via an AGI-Script. When parsing the AGI-Variables I can see one that look like that: [agi_channel] => DAHDI/i3/211123456-89c What hat do the values mean in detail, please? DAHDI : this is clear i3 : does it mean, that the call comes in via E1-Port 3? 211123456 : Incoming-Call Caller-ID -89c : ? WANPIPE Release: 7.0.1 DAHDI Version: 2.6.2 Echo Canceller: HWEC libpri version: 1.4.12 Best regards -Thorsten- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming Fax to Recipient using OCR
On Tue, Nov 6, 2012 at 1:50 PM, Roy Abshire wrote: > I have fax working but since most people and services don't know how to > Fax to Extensions, > I installed tesseract to convert the Fax to Text. > > I really only need the First Page converted and will tell Faxers to make > sure they put To: Name on the cover page. > > tesseract is converting the entire fax fine but its unnecessary and extra > time to convert the entire fax. > > I searched and can't find anything on how to tell it just to do the first > page. Does anyone have any ideas? If you're passing a TIFF file to tesseract, you can pass it through imagemagick first to pop off the first "page". This really seems off-topic for Asterisk. -- -Chris Harrington ACSDi Office: 763.559.5800 Mobile Phone: 612.326.4248 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming Fax to Recipient using OCR
If ($_ =~ /[Tt][Oo]\:.[Nn]ame/) { Is the way I do it. If ($_ =~ /[Tt][Oo]..[Nn]ame/) { Would also work -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Roy Abshire Sent: Tuesday, November 06, 2012 1:51 PM To: Asterisk Users Subject: [asterisk-users] Incoming Fax to Recipient using OCR I have fax working but since most people and services don't know how to Fax to Extensions, I installed tesseract to convert the Fax to Text. I really only need the First Page converted and will tell Faxers to make sure they put To: Name on the cover page. tesseract is converting the entire fax fine but its unnecessary and extra time to convert the entire fax. I searched and can't find anything on how to tell it just to do the first page. Does anyone have any ideas? I created a perl script I borrowed but I don't know PERL. I know PHP so can someone show me how to use REGEX in Perl to search the output.txt file for the to: name or TO: NAME or To: Name Then I want to do something like: Switch($to) { Case: "Roy" -> Email u...@gmail.com Case: "Jeff" -> Email u...@yahoo.com Default: Email ad...@domain.com } -- Co-op Vacation Rentals www.coopvr.com 15218 Summit Ave Suite #300-354 Fontana, CA 92336 Phone/Fax (855) 760-COOP (2667) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Incoming Fax to Recipient using OCR
I have fax working but since most people and services don't know how to Fax to Extensions, I installed tesseract to convert the Fax to Text. I really only need the First Page converted and will tell Faxers to make sure they put To: Name on the cover page. tesseract is converting the entire fax fine but its unnecessary and extra time to convert the entire fax. I searched and can't find anything on how to tell it just to do the first page. Does anyone have any ideas? I created a perl script I borrowed but I don't know PERL. I know PHP so can someone show me how to use REGEX in Perl to search the output.txt file for the to: name or TO: NAME or To: Name Then I want to do something like: Switch($to) { Case: "Roy" -> Email u...@gmail.com Case: "Jeff" -> Email u...@yahoo.com Default: Email ad...@domain.com } -- Co-op Vacation Rentals www.coopvr.com 15218 Summit Ave Suite #300-354 Fontana, CA 92336 Phone/Fax (855) 760-COOP (2667) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Incoming Fax to Recipient using OCR
I have fax working but since most people and services don't know how to Fax to Extensions, I installed tesseract to convert the Fax to Text. I really only need the First Page converted and will tell Faxers to make sure they put To: Name on the cover page. tesseract is converting the entire fax fine but its unnecessary and extra time to convert the entire fax. I searched and can't find anything on how to tell it just to do the first page. Does anyone have any ideas? I created a perl script I borrowed but I don't know PERL. I know PHP so can someone show me how to use REGEX in Perl to search the output.txt file for the to: name or TO: NAME or To: Name Then I want to do something like: Switch($to) { Case: "Roy" -> Email u...@gmail.com Case: "Jeff" -> Email u...@yahoo.com Default: Email ad...@domain.com } -- Co-op Vacation Rentals www.coopvr.com 15218 Summit Ave Suite #300-354 Fontana, CA 92336 Phone/Fax (855) 760-COOP (2667) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming SIP call is rejected always.
- Original Message - > From: "Yaroslav Panych" > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > Sent: Tuesday, April 17, 2012 6:56:17 PM > Subject: Re: [asterisk-users] Incoming SIP call is rejected always. > > 2012/4/18 Matthew Jordan : > > I imagine that this is the case, as ASTERISK-19601 noted that > > when this situation occurs, the NOTICE message indicates that > > there is a failure to match the extension, as opposed to a failure > > to match an allowed domain. > > Yes, it was hell to detect real error cause(I was forced to learn how > to debug in KDevelop in less than four hours). Yes, it looks like > ASTERISK-19601. But still I cannot understand why asterisk extracts > wrong domain from request. > > However, in your SIP configuration you have set > > allowexternaldomains to no. > Yes, it is intended. > > > Without knowing the URI the INVITE request was addressed to, its > > difficult to say what might be the actual cause of this. > I first letter I have provided CLI log which contains full request > packets(Authless and authed INVITE included). > > Probably I do not understand how to configure Asterisk: > I have one asterisk. It serves SIP domain example.com. This asterisk > must be able to establish session with registered client of this > account and also must be able to accept incoming sessions. No > sessions > with 3rd-party accounts on 3rd-party domains allowed to established. > How I should setup this asterisk? Well, I can't tell you how to configure your Asterisk server. However, I can tell you why Asterisk rejected the INVITE request. The URI that the INVITE request was addressed to is 4001020@192.168.8.2:5060. The domain portion of this URI is 192.168.8.2. Hence, the allowed domains need to include that particular IPv4 address. Looking at the allowed domains you've specified in sip.conf, we have: domain=sop-korniychuk domain=192.168.8.1 domain=192.168.8.1:5062 So, since the INVITE request does not match any of those three domains, its rejected. Note: I noticed that you have autodomain set to yes; I'm going to assume that the IPv4 address 192.168.8.2 is not associated with the server. Matt > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: >http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming SIP call is rejected always.
2012/4/18 Matthew Jordan : > I imagine that this is the case, as ASTERISK-19601 noted that > when this situation occurs, the NOTICE message indicates that > there is a failure to match the extension, as opposed to a failure > to match an allowed domain. Yes, it was hell to detect real error cause(I was forced to learn how to debug in KDevelop in less than four hours). Yes, it looks like ASTERISK-19601. But still I cannot understand why asterisk extracts wrong domain from request. > However, in your SIP configuration you have set allowexternaldomains to no. Yes, it is intended. > Without knowing the URI the INVITE request was addressed to, its > difficult to say what might be the actual cause of this. I first letter I have provided CLI log which contains full request packets(Authless and authed INVITE included). Probably I do not understand how to configure Asterisk: I have one asterisk. It serves SIP domain example.com. This asterisk must be able to establish session with registered client of this account and also must be able to accept incoming sessions. No sessions with 3rd-party accounts on 3rd-party domains allowed to established. How I should setup this asterisk? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming SIP call is rejected always.
Without knowing the URI the INVITE request was addressed to, its difficult to say what might be the actual cause of this. However, in your SIP configuration you have set allowexternaldomains to no. That implies that if the domain of the URI does not match any of the allowed domains you have set, that the INVITE request will be rejected. I imagine that this is the case, as ASTERISK-19601 noted that when this situation occurs, the NOTICE message indicates that there is a failure to match the extension, as opposed to a failure to match an allowed domain. Matthew Jordan Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com & http://asterisk.org - Original Message - > From: "Yaroslav Panych" > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > Sent: Tuesday, April 17, 2012 4:58:14 PM > Subject: Re: [asterisk-users] Incoming SIP call is rejected always. > > 2012/4/17 Danny Nicholas : > > Maybe it needs to be _4001020? > > > > Not, it doesn't. Actually I have traced this incoming call step by > step. Real reason it refuses - wrong domain. But why it wrong - have > not any idea. > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: >http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming SIP call is rejected always.
2012/4/17 Danny Nicholas : > Maybe it needs to be _4001020? > Not, it doesn't. Actually I have traced this incoming call step by step. Real reason it refuses - wrong domain. But why it wrong - have not any idea. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming SIP call is rejected always.
Maybe it needs to be _4001020? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Yaroslav Panych Sent: Tuesday, April 17, 2012 7:38 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Incoming SIP call is rejected always. Hi Have an asterisk. Setup a couple of friends. Sip.conf - http://pastebin.com/zUgiYbBi Trying to make incoming call, and have such error(cli output) http://pastebin.com/zFfgYcNR NOTICE[4994]: chan_sip.c:23316 handle_request_invite: Call from 'RMT20' (192.168.8.1:5062) to extension '4001020' rejected because extension not found in context 'rmt-context'. But, as you see, there is such extension. What I'm doing wrong? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Incoming SIP call is rejected always.
Hi Have an asterisk. Setup a couple of friends. Sip.conf - http://pastebin.com/zUgiYbBi Trying to make incoming call, and have such error(cli output) http://pastebin.com/zFfgYcNR NOTICE[4994]: chan_sip.c:23316 handle_request_invite: Call from 'RMT20' (192.168.8.1:5062) to extension '4001020' rejected because extension not found in context 'rmt-context'. But, as you see, there is such extension. What I'm doing wrong? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming Call Recording
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Rick Hall Sent: Friday, June 10, 2011 3:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Incoming Call Recording Longtime lurker, first time poster. :) A client of mine is in need of having Asterisk record every call that comes in from a specific incoming route. I've added the following lines to the sip_additional.conf file, but no recordings are showing up in the /var/spool/asterisk/monitor/ folder. record_out=always record_in=always Another page I came across on Google (http://www.voip-info.org/wiki/view/Asterisk+cmd+Monitor) suggested I add the following line to my sip.conf file: exten => 2060,3,Monitor(wav,myfilename) I can see how this could work, but I'm not sure what to replace "2060" with, as what I need setup is the record of all incoming calls across the board, not just calls associated with a particular extension number (ie: 2060). Your sure is appreciated! -- Rick Hall Senior Vice President ReadyWire Multimedia Solutions Affordable Website & Reseller Hosting http://www.readywire.com/ (312) 278-4446 x5446 Technical Support: 24 hours a day / 7 days a week Customer Login...: https://secure.readywire.com/ Server Notices.: http://status.readywire.com/ Support Center: https://secure.readywire.com/ Twitter.: http://twitter.com/readywire Blog: http://blog.readywire.com/ This message contains confidential information and is intended only for the individual named. If you are not the named addressee you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately by e-mail if you have received this e-mail by mistake and delete this e-mail from your system. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. ReadyWire Multimedia Solutions. PO BOX 811061, Chicago, IL, USA, 60681. www.readywire.com. This will do the trick, but you should play the "you are being recorded" file to cover your assets exten => s,n,MixMonitor(Zap_${UNIQUEID}.wav|av(0}V(0)) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Incoming Call Recording
Longtime lurker, first time poster. :) A client of mine is in need of having Asterisk record every call that comes in from a specific incoming route. I've added the following lines to the sip_additional.conf file, but no recordings are showing up in the /var/spool/asterisk/monitor/ folder. record_out=always record_in=always Another page I came across on Google ( http://www.voip-info.org/wiki/view/Asterisk+cmd+Monitor) suggested I add the following line to my sip.conf file: exten => 2060,3,Monitor(wav,myfilename) I can see how this could work, but I'm not sure what to replace "2060" with, as what I need setup is the record of all incoming calls across the board, not just calls associated with a particular extension number (ie: 2060). Your sure is appreciated! -- Rick Hall Senior Vice President ReadyWire Multimedia Solutions Affordable Website & Reseller Hosting http://www.readywire.com/ (312) 278-4446 x5446 Technical Support: 24 hours a day / 7 days a week Customer Login...: https://secure.readywire.com/ Server Notices.: http://status.readywire.com/ Support Center: https://secure.readywire.com/ Twitter.: http://twitter.com/readywire Blog: http://blog.readywire.com/ This message contains confidential information and is intended only for the individual named. If you are not the named addressee you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately by e-mail if you have received this e-mail by mistake and delete this e-mail from your system. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. ReadyWire Multimedia Solutions. PO BOX 811061, Chicago, IL, USA, 60681. www.readywire.com. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Incoming SRTP call not working with Bria iPhone Edition
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi Everybody, I am experiencing some troubles with my Bria iPhone Edition (v. 1.2.8 build 5312, on iOS 4.2.1 iPhone 3G) and Asterisk 1.8.3.2 + TLS/SRTP on LAN (without NAT). With 2 computer clients (Blink, one on Mac, one on Windows/Linux),9i can have a very fine secure conversation in both directions. When I want to do the same with my iPhone, only outgoing calls are working. If i try to call (from Blink Win/Mac) my iPhone, Bria is not ringing. Asterisk logs only said that nobody has picked up : {{{ == Using SIP RTP CoS mark 5 -- Executing [400@local:1] Dial("SIP/500-0004", "SIP/400,20") in new stack == Using SIP RTP CoS mark 5 -- Called 400 SSL certificate ok -- Nobody picked up in 2 ms }}} My config files are : * sip.conf : {{{ tlsenable=yes tlsbindaddr=0.0.0.0 tlscertfile=/etc/asterisk/keys/asterisk.pem tlscafile=/etc/asterisk/keys/ca.crt tlscipher=ALL tlsclientmethod=tlsv1 ;none of the others seem to work with Blink as the client [400] type=peer secret=400 ;note that this is NOT a secure password host=dynamic context=local dtmfmode=rfc2833 disallow=all allow=g722,gsm transport=tls encryption=yes context=local [500] type=peer secret=500 ;note that this is NOT a secure password host=dynamic context=local dtmfmode=rfc2833 disallow=all allow=g722,gsm transport=tls encryption=yes context=local }}} * extensions.conf : {{{ exten => 400,1,Dial(SIP/400,20) exten => 400,2,VoiceMail(u400@default) exten => 400,VoiceMail(b400@default) exten => 400,3,Hangup() exten => 500,1,Dial(SIP/500,20) exten => 500,2,VoiceMail(u500@default) exten => 500,VoiceMail(b500@default) exten => 500,3,Hangup() }}} If I try with a simple SIP (no TLS/SRTP) configuration, the iPhone is ringing and I can pick up but there is no sound. It is working fine on the other direction. Network Traversal Strategy is set to "Server Managed" (I have tried the others with success). I have already ask for CounterPath/Bria support, but I didn't have a positive answer yet. What is wrong with my settings? Thanks for your help. - -- Alexis de BRUYN email : ale...@de-bruyn.fr -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.10 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iEYEARECAAYFAk2VrlAACgkQNy3UyEOc6xUCDwCfVvGO2l80LAJZMn1T4+1UIzcj ZN8AoJC4o7R6FkrN7jZ2q48hDAWca9nv =y7JN -END PGP SIGNATURE- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] incoming
On Mon, 3 Jan 2011, Thomas Perron wrote: So, one Asterisk machine handling up to 100 DID numbers, correct? The number of DIDs is not limited. You could handle a bazillion DIDs with a simple dial plan like: exten = _!.,1, verbose(1,[${ext...@${context}]) exten = _!.,n, playback(demo-congrats) exten = _!.,n, hangup() I assume that the DID mumbers dialed would be the exaxt match needed to start the respective context. Correct? The exten does not determine which context is started. The provider configuration does. Matching is facilitated by patterns. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] incoming
On Mon, Jan 03, 2011 at 02:41:36AM +0400, Thomas Perron wrote: >Cool. So, one Asterisk machine handling up to 100 DID numbers, correct? As many as you like, modulo memory and CPU requirements. >I assume that the DID mumbers dialed would be the exaxt match needed >to start the respective context. Correct? Depends on how they're presented to you by the DID provider. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] incoming
On Sun, 2 Jan 2011, Thomas Perron wrote: Is it possible to have Calls incoming to different DIDs? Yes*, depending on whether your provider 'provides' the DID in the call setup. *) Better subjects attract more readers. More detail yields better answers. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] incoming
Cool. So, one Asterisk machine handling up to 100 DID numbers, correct? Yes. I will have unique IVR flows/plans for each. I assume that the DID mumbers dialed would be the exaxt match needed to start the respective context. Correct? On 1/3/11, Rick Hall wrote: > Yes, I don't see why not. You just need to setup an IVR for each business > and then assign each individual DID to the appropriate IVR. > > This may help: > > http://www.voip-info.org/wiki/view/Asterisk+tips+ivr+menu > > Cheers! > > Rick > > -- > Rick Hall > Senior Vice President > ReadyWire Multimedia Solutions > > Affordable Website & Reseller Hosting > http://www.readywire.com/ > (312) 278-4446 x5446 > > Technical Support: > 24 hours a day / 7 days a week > > Customer Login...: https://secure.readywire.com/ > Server Notices.: http://status.readywire.com/ > Support Center: https://secure.readywire.com/ > Twitter.: http://twitter.com/readywire > Blog: http://blog.readywire.com/ > > This message contains confidential information and is intended only for the > individual named. If you are not the named addressee you should not > disseminate, distribute or copy this e-mail. Please notify the sender > immediately by e-mail if you have received this e-mail by mistake and delete > this e-mail from your system. E-mail transmission cannot be guaranteed to be > secure or error-free as information could be intercepted, corrupted, lost, > destroyed, arrive late or incomplete, or contain viruses. The sender > therefore does not accept liability for any errors or omissions in the > contents of this message, which arise as a result of e-mail transmission. If > verification is required please request a hard-copy version. ReadyWire > Multimedia Solutions. PO BOX 811061, Chicago, IL, USA, 60681. > www.readywire.com. > > > > On Sun, Jan 2, 2011 at 11:50 AM, Thomas Perron > wrote: > >> Is it possible to have >> Calls incoming to different DIDs? >> I want an AA that handles 100s of businesses. >> >> [Incoming-pizza] >> Exten => 4045551212,1,Goto(pizza,s,1) >> >> [Incoming-hvac] >> Exten => 8085551212,1,Goto(hvac,s,1) >> >> [Incoming-gutter] >> Exten => 6175551212,1,Goto(gutter,s,1) >> >> -- >> _ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] incoming
Yes, I don't see why not. You just need to setup an IVR for each business and then assign each individual DID to the appropriate IVR. This may help: http://www.voip-info.org/wiki/view/Asterisk+tips+ivr+menu Cheers! Rick -- Rick Hall Senior Vice President ReadyWire Multimedia Solutions Affordable Website & Reseller Hosting http://www.readywire.com/ (312) 278-4446 x5446 Technical Support: 24 hours a day / 7 days a week Customer Login...: https://secure.readywire.com/ Server Notices.: http://status.readywire.com/ Support Center: https://secure.readywire.com/ Twitter.: http://twitter.com/readywire Blog: http://blog.readywire.com/ This message contains confidential information and is intended only for the individual named. If you are not the named addressee you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately by e-mail if you have received this e-mail by mistake and delete this e-mail from your system. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. ReadyWire Multimedia Solutions. PO BOX 811061, Chicago, IL, USA, 60681. www.readywire.com. On Sun, Jan 2, 2011 at 11:50 AM, Thomas Perron wrote: > Is it possible to have > Calls incoming to different DIDs? > I want an AA that handles 100s of businesses. > > [Incoming-pizza] > Exten => 4045551212,1,Goto(pizza,s,1) > > [Incoming-hvac] > Exten => 8085551212,1,Goto(hvac,s,1) > > [Incoming-gutter] > Exten => 6175551212,1,Goto(gutter,s,1) > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] incoming
Is it possible to have Calls incoming to different DIDs? I want an AA that handles 100s of businesses. [Incoming-pizza] Exten => 4045551212,1,Goto(pizza,s,1) [Incoming-hvac] Exten => 8085551212,1,Goto(hvac,s,1) [Incoming-gutter] Exten => 6175551212,1,Goto(gutter,s,1) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming calls through SS7 for data modem transmissions - possible??
Matt, We are located on Costa Rica and so far there's just 1 TELCO running the industrym with the CAFTA treatment the carrier had to open for interconnection but they get to define the ground rules for the interconnection. They are arguing ISDN is and "end customer" circuit and you cannot use it to "resell" service. You need an "Interconnect" circuit, to resell dial up access, and they require you to support SS7 for "Interconnect" T1s It's just the carrier raising the bar waiting for 90% of the competition to drop off, and a little porcentage to make it. We asked for quotes for multiple solutions, most of them ranging 15 000$, and finally come to the conclusion Asterisk and SS7 would be the only viable option for this project. Cheers, On Tue, Nov 30, 2010 at 9:21 AM, Matt Watson wrote: > Just out of curiosity, what country are you in? > > I agree with the others in this thread, this seems very bizzare that the > telco requires you to do SS7 for dialup connections. I would ask them for > specifics about the "legal" issues with what you are doing - it sounds to me > like they are just trying to upsell you on a more expensive product. > > I am in Canada and we run exactly the configuration you are currently > doing... we still have dialup internet customers that dial into AS5300's via > PRI's. Our telco has a PRI product gear specifically for this use... they > call it 'ISP-PRI' I'm not entirely sure what the restriction is on it I > have also just kind of assumed that it is inbound calls only, but I've never > tried making outbound calls on them. I do know they 25-30% cheaper than our > regular voice PRIs though. > > -- > Matt > > > 2010/11/24 José Pablo Méndez Soto > > Hello, >> >> We are working on implementing a solution for a medium service provider. >> They were previously using a Cisco AS5300 gateway with some PRI trunks to >> receive modem calls, then route them out the Internet. >> >> The Telco they were buying the trunks to discovered this configuration and >> restricted them due to legal conventions, and stated that in order to >> continue doing this, they would have to talk SS7 directly. >> >> > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- Robert -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming calls through SS7 for data modem transmissions - possible??
Just out of curiosity, what country are you in? I agree with the others in this thread, this seems very bizzare that the telco requires you to do SS7 for dialup connections. I would ask them for specifics about the "legal" issues with what you are doing - it sounds to me like they are just trying to upsell you on a more expensive product. I am in Canada and we run exactly the configuration you are currently doing... we still have dialup internet customers that dial into AS5300's via PRI's. Our telco has a PRI product gear specifically for this use... they call it 'ISP-PRI' I'm not entirely sure what the restriction is on it I have also just kind of assumed that it is inbound calls only, but I've never tried making outbound calls on them. I do know they 25-30% cheaper than our regular voice PRIs though. -- Matt 2010/11/24 José Pablo Méndez Soto > Hello, > > We are working on implementing a solution for a medium service provider. > They were previously using a Cisco AS5300 gateway with some PRI trunks to > receive modem calls, then route them out the Internet. > > The Telco they were buying the trunks to discovered this configuration and > restricted them due to legal conventions, and stated that in order to > continue doing this, they would have to talk SS7 directly. > > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming calls through SS7 for datamodemtransmissions - possible??
Thanks Cary, The first topology we are working on should be the best way then. Asterisk will answer SS7 calls, route them to the ISDN channels to be terminated by the AS5300 as they were doing before. I think TDM-2-TDM shouldn't be that much of a problem eh? No further equipment needed? *José Pablo Méndez * 2010/11/24 Cary Fitch > I understand the problem. You can’t resell PRI connections. > > > > I don’t think Asterisk can convert TDM to IP. It does convert TDM to SIP > which is then sent out over IP.What you want to do is have it do the > TDM/Modem conversion without the PRIs and Cisco Gear. > > > > There used to be a way to do this, and maybe still is but not just with > Asterisk perhaps. > > > > I know that Ascend/Lucent TNTs (and I am sure some other equipment) could > take TDM trunks, which could be SS7 trunks, and convert them to IP. > > > > The point in this is that they are SS7 based. You can take SS7 trunks from > either the Asterisk box or direct from the Telco and convert them to IP. > > > > NO PRIs involved. Yes, more “telco grade carrier equipment” but no PRIs. > > > > A lot of this equipment was available by the pound a few years back. > > > > Cary > > > -- > > *From:* José Pablo Méndez Soto [mailto:aux...@gmail.com] > *Sent:* Wednesday, November 24, 2010 8:34 PM > *To:* Asterisk Users Mailing List - Non-Commercial Discussion > *Cc:* ca...@usawide.net > *Subject:* Re: [asterisk-users] Incoming calls through SS7 for > datamodemtransmissions - possible?? > > > > Thanks Cary, > > What happens is, the Telco won't allow the small company to resell the ISDN > connections, meaning, they bought the trunks and DIDs, then sold dialing > plans to route incoming calls through the PRIs out the Internet. This is not > the issue though. We definitely have to migrate to an SS7 capable platform, > because that is the only way the Telco allows them to resell the dial-up > connections (not ISDN), and Asterisk is the current bet. > > If we can get Asterisk to pick up those calls via SS7, then authenticate > them, send them out to the Internet, we would be achieving a %100 usage on > the Digium cards, because one of them wouldn't be used to talk to the AS. > > Can Asterisk do this? > > > Thanks again, > > *José Pablo Méndez** >* > > On Wed, Nov 24, 2010 at 7:59 PM, Cary Fitch wrote: > > I am not sure where you are and what legal conventions are involved. > > > > Are you saying the Telco (and legal restrictions) say you can’t send calls > to the internet via the AS5300 but you can if Asterisk does it directly? > What is the “logic” in that? > > > > Or are they saying your Telco to Asterisk trunks have to be SS7 controlled? > > > > > Or are you concerned about Asterisk handling the TDM to IP conversion in an > adequate manner? > > > > I am not sure/aware myself that Asterisk will do a modem to IP conversion. > I think in your example the AS5300 is doing that. > > > > What is the Telco’s problem in doing what the customer was doing before? > > > > Feel free to correspond directly if you want to. > > > > Cary Fitch > > > -- > > *From:* asterisk-users-boun...@lists.digium.com [mailto: > asterisk-users-boun...@lists.digium.com] *On Behalf Of *José Pablo Méndez > Soto > *Sent:* Wednesday, November 24, 2010 7:31 PM > *To:* asterisk-users@lists.digium.com > *Subject:* [asterisk-users] Incoming calls through SS7 for data > modemtransmissions - possible?? > > > > Hello, > > We are working on implementing a solution for a medium service provider. > They were previously using a Cisco AS5300 gateway with some PRI trunks to > receive modem calls, then route them out the Internet. > > The Telco they were buying the trunks to discovered this configuration and > restricted them due to legal conventions, and stated that in order to > continue doing this, they would have to talk SS7 directly. > > We are planning on solving this by placing an Asterisk server with some > TE410 cards talking SS7 to Telco, and another 4 ISDN ports talking to the > AS5300 for the dial-up to complete after authenticating against a RADIUS > server. > > My questions is: can we use only Asterisk to complete/terminate the dial-up > connection, removing the AS5300 out of the picture? > > Current topology to be set-up: > Telco --> SS7 --> TE410P-AsteriskServer --> ISDN --> AS5300 --> Internet > > Ideal topology: > Telco --> SS7 --> TE410P-AsteriskServer --> Internet > > > Some posts talk
Re: [asterisk-users] Incoming calls through SS7 for data modemtransmissions - possible??
Thanks Cary, What happens is, the Telco won't allow the small company to resell the ISDN connections, meaning, they bought the trunks and DIDs, then sold dialing plans to route incoming calls through the PRIs out the Internet. This is not the issue though. We definitely have to migrate to an SS7 capable platform, because that is the only way the Telco allows them to resell the dial-up connections (not ISDN), and Asterisk is the current bet. If we can get Asterisk to pick up those calls via SS7, then authenticate them, send them out to the Internet, we would be achieving a %100 usage on the Digium cards, because one of them wouldn't be used to talk to the AS. Can Asterisk do this? Thanks again, *José Pablo Méndez * On Wed, Nov 24, 2010 at 7:59 PM, Cary Fitch wrote: > I am not sure where you are and what legal conventions are involved. > > > > Are you saying the Telco (and legal restrictions) say you can’t send calls > to the internet via the AS5300 but you can if Asterisk does it directly? > What is the “logic” in that? > > > > Or are they saying your Telco to Asterisk trunks have to be SS7 controlled? > > > > > Or are you concerned about Asterisk handling the TDM to IP conversion in an > adequate manner? > > > > I am not sure/aware myself that Asterisk will do a modem to IP conversion. > I think in your example the AS5300 is doing that. > > > > What is the Telco’s problem in doing what the customer was doing before? > > > > Feel free to correspond directly if you want to. > > > > Cary Fitch > > > -- > > *From:* asterisk-users-boun...@lists.digium.com [mailto: > asterisk-users-boun...@lists.digium.com] *On Behalf Of *José Pablo Méndez > Soto > *Sent:* Wednesday, November 24, 2010 7:31 PM > *To:* asterisk-users@lists.digium.com > *Subject:* [asterisk-users] Incoming calls through SS7 for data > modemtransmissions - possible?? > > > > Hello, > > We are working on implementing a solution for a medium service provider. > They were previously using a Cisco AS5300 gateway with some PRI trunks to > receive modem calls, then route them out the Internet. > > The Telco they were buying the trunks to discovered this configuration and > restricted them due to legal conventions, and stated that in order to > continue doing this, they would have to talk SS7 directly. > > We are planning on solving this by placing an Asterisk server with some > TE410 cards talking SS7 to Telco, and another 4 ISDN ports talking to the > AS5300 for the dial-up to complete after authenticating against a RADIUS > server. > > My questions is: can we use only Asterisk to complete/terminate the dial-up > connection, removing the AS5300 out of the picture? > > Current topology to be set-up: > Telco --> SS7 --> TE410P-AsteriskServer --> ISDN --> AS5300 --> Internet > > Ideal topology: > Telco --> SS7 --> TE410P-AsteriskServer --> Internet > > > Some posts talk about zapRAS being able to accomplish this, not quite sure > though > > Sounds like possible: > http://lists.digium.com/pipermail/asterisk-users/2004-January/026956.html > > Sounds like not possible: > http://lists.digium.com/pipermail/asterisk-users/2009-November/240202.html > > > Thanks in advance, > > > *José Pablo Méndez** > * > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming calls through SS7 for data modemtransmissions - possible??
I am not sure where you are and what legal conventions are involved. Are you saying the Telco (and legal restrictions) say you cant send calls to the internet via the AS5300 but you can if Asterisk does it directly? What is the logic in that? Or are they saying your Telco to Asterisk trunks have to be SS7 controlled? Or are you concerned about Asterisk handling the TDM to IP conversion in an adequate manner? I am not sure/aware myself that Asterisk will do a modem to IP conversion. I think in your example the AS5300 is doing that. What is the Telcos problem in doing what the customer was doing before? Feel free to correspond directly if you want to. Cary Fitch _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of José Pablo Méndez Soto Sent: Wednesday, November 24, 2010 7:31 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Incoming calls through SS7 for data modemtransmissions - possible?? Hello, We are working on implementing a solution for a medium service provider. They were previously using a Cisco AS5300 gateway with some PRI trunks to receive modem calls, then route them out the Internet. The Telco they were buying the trunks to discovered this configuration and restricted them due to legal conventions, and stated that in order to continue doing this, they would have to talk SS7 directly. We are planning on solving this by placing an Asterisk server with some TE410 cards talking SS7 to Telco, and another 4 ISDN ports talking to the AS5300 for the dial-up to complete after authenticating against a RADIUS server. My questions is: can we use only Asterisk to complete/terminate the dial-up connection, removing the AS5300 out of the picture? Current topology to be set-up: Telco --> SS7 --> TE410P-AsteriskServer --> ISDN --> AS5300 --> Internet Ideal topology: Telco --> SS7 --> TE410P-AsteriskServer --> Internet Some posts talk about zapRAS being able to accomplish this, not quite sure though Sounds like possible: http://lists.digium.com/pipermail/asterisk-users/2004-January/026956.html <mailto:asterisk-users@lists.digium.com> Sounds like not possible: http://lists.digium.com/pipermail/asterisk-users/2009-November/240202.html Thanks in advance, José Pablo Méndez -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Incoming calls through SS7 for data modem transmissions - possible??
Hello, We are working on implementing a solution for a medium service provider. They were previously using a Cisco AS5300 gateway with some PRI trunks to receive modem calls, then route them out the Internet. The Telco they were buying the trunks to discovered this configuration and restricted them due to legal conventions, and stated that in order to continue doing this, they would have to talk SS7 directly. We are planning on solving this by placing an Asterisk server with some TE410 cards talking SS7 to Telco, and another 4 ISDN ports talking to the AS5300 for the dial-up to complete after authenticating against a RADIUS server. My questions is: can we use only Asterisk to complete/terminate the dial-up connection, removing the AS5300 out of the picture? Current topology to be set-up: Telco --> SS7 --> TE410P-AsteriskServer --> ISDN --> AS5300 --> Internet Ideal topology: Telco --> SS7 --> TE410P-AsteriskServer --> Internet Some posts talk about zapRAS being able to accomplish this, not quite sure though Sounds like possible: http://lists.digium.com/pipermail/asterisk-users/2004-January/026956.html Sounds like not possible: http://lists.digium.com/pipermail/asterisk-users/2009-November/240202.html Thanks in advance, *José Pablo Méndez * -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming calls
On Thu, 18 Nov 2010, Flavio Miranda wrote: Looking to dahdi show channles , I realized that all the trunks was in the same context. So, I have changed this and everything works! That's why I prefer to work from what Asterisk parsed the file as, not what the poster thinks :) -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming calls
Hi Steve, thanks for the tips "Better bait = better fish" ! As you said, I am in the right track. Looking to dahdi show channles , I realized that all the trunks was in the same context. So, I have changed this and everything works! thanks you !! Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda > Date: Thu, 18 Nov 2010 11:53:26 -0800 > From: asterisk@sedwards.com > To: asterisk-users@lists.digium.com > Subject: Re: [asterisk-users] Incoming calls > > On Thu, 18 Nov 2010, Flavio Miranda wrote: > > > I'd like that each analog trunk of my TDM410p was received in different > > extension. So, in dahdi-channel.conf and chan-dahdi.conf I put each > > trunk in a different context and in my extensions.conf, under [default] > > I put such contexts and an especific estension to answer it. therefore, > > when I get call, it always is ringing on the first extensions, dont > > matter trunk. Anybody could teach me how can I organize that ? > > 0) Use a subject that gives a clue what you're looking for. Almost > everybody has had a question about an incomig call at some point in time. > Better bait = better fish. > > 1) It sounds like you have a clue about how to do it and are on the right > track. > > 2) Including some details like the console output from: > > zap show channel 1 (I'm a 1.2 Luddite.) > zap show channel 2 > zap show channel 3 > zap show channel 4 > > as well as the console log from a call coming in on each channel > > will help in assisting you in resolving this issue. > > -- > Thanks in advance, > - > Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST > Newline Fax: +1-760-731-3000 > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: >http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming calls
On Thu, 18 Nov 2010, Flavio Miranda wrote: > I'd like that each analog trunk of my TDM410p was received in different > extension. So, in dahdi-channel.conf and chan-dahdi.conf I put each > trunk in a different context and in my extensions.conf, under [default] > I put such contexts and an especific estension to answer it. therefore, > when I get call, it always is ringing on the first extensions, dont > matter trunk. Anybody could teach me how can I organize that ? 0) Use a subject that gives a clue what you're looking for. Almost everybody has had a question about an incomig call at some point in time. Better bait = better fish. 1) It sounds like you have a clue about how to do it and are on the right track. 2) Including some details like the console output from: zap show channel 1 (I'm a 1.2 Luddite.) zap show channel 2 zap show channel 3 zap show channel 4 as well as the console log from a call coming in on each channel will help in assisting you in resolving this issue. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Incoming calls
Hi all, I'd like that each analog trunk of my TDM410p was received in different extension. So, in dahdi-channel.conf and chan-dahdi.conf I put each trunk in a different context and in my extensions.conf, under [default] I put such contexts and an especific estension to answer it. therefore, when I get call, it always is ringing on the first extensions, dont matter trunk . Anybody could teach me how can I organize that ? Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Incoming calls
Hi all, After a lot of trouble with a TE110p working with mfcr2 , brazil variant, everything looks great,but I can not go out of my calls. When I try I receive the following log: == Using SIP RTP CoS mark 5-- Executing [33220...@local:1] Dial("SIP/4804-001a", "DAHDI/g11/33220567,,T") in new stack == Everyone is busy/congested at this time (1:0/1/0)-- Auto fallthrough, channel 'SIP/4804-001a' status is 'CONGESTION'This is my dahdi show status:Digium Wildcard TE110P T1/E1 Card 0 REC 0 0 0 CAS HDB3 CRC4 0 db (CSU)/0-133 feet (DSX-1)thi´s my dahdi show channels: asterisk*CLI> dahdi show channels Chan Extension Context Language MOH InterpretBlockedState pseudodefault default In Service 1 4800 default default In Service 2 4800 defaultdefault In Service 3 4805 defaultdefault In Service 4defaultdefault In Service 5defaultdefault In Service 6defaultdefault In Service 7default default In Service 8default default In Service 9default default In Service 10 defaultdefault In Service 11 defaultdefault In Service 12defaultdefault In Service 13defaultdefault In Service 14defaultdefault In Service 15defaultdefault In Service 17default default In Service 18default default In Service 19default default In Service 20 defaultdefault In Service 21 defaultdefault In Service 22defaultdefault In Service 23defaultdefault In Service 24defaultdefault In Service 25defaultdefault In Service 26default default In Service 27default default In Service 28default default In Service 29 defaultdefault In Service 30 defaultdefault In Service 31defaultdefault In Service *In my incoming call , the log is: MFC/R2 call offered on chan 1. ANI = 1221341400, DNIS = 4804, Category = National SubscriberNew MFC/R2 call detected on chan 2. and don't ring nowhere! Thanks for help!Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] incoming call FXO
Ok. Problem solved . Thank you very much!!! Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda Date: Wed, 15 Sep 2010 09:56:36 -0400 From: zisha...@gmail.com To: kpflem...@digium.com; asterisk-users@lists.digium.com Subject: Re: [asterisk-users] incoming call FXO As Kevin said, you need to define an 's' extension where the calls will be answered. Seems like you are using default configuration. Open file 'extensions.conf' in /etc/asterisk folder and look for context named [default]. If it is not there, create one and add something under it, e.g., [default] exten => s,1,Verbose( - - - Call received - - - ) exten => s,n,Playback(hello-world) extent => s,n,HangUp() Then do a 'core extensions reload' on Asterisk CLI. Now calling in on FXO should play the message 'hello-world' (assuming this sound file exists in the sound folder of asterisk), and you'll see the call activity on the CLI. For the rest, you need to consult chapters 5 and 6 of 'Asterisk-The Future of Telephony' book. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-09-15 8:59 AM, "Kevin P. Fleming" wrote: On 09/15/2010 07:20 AM, Flavio Miranda wrote: > Recently I have instaled one Digium TDM410 on my... Right, that's what the message is telling you. For incoming calls on FXO, they can *only* be sent to the 's' extension in the target context, since there is no target number passed over the FXO connection. You'll have to create an 's' extension to handle incoming calls however you like. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] incoming call FXO
As Kevin said, you need to define an 's' extension where the calls will be answered. Seems like you are using default configuration. Open file 'extensions.conf' in /etc/asterisk folder and look for context named [default]. If it is not there, create one and add something under it, e.g., [default] exten => s,1,Verbose( - - - Call received - - - ) exten => s,n,Playback(hello-world) extent => s,n,HangUp() Then do a 'core extensions reload' on Asterisk CLI. Now calling in on FXO should play the message 'hello-world' (assuming this sound file exists in the sound folder of asterisk), and you'll see the call activity on the CLI. For the rest, you need to consult chapters 5 and 6 of 'Asterisk-The Future of Telephony' book. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-09-15 8:59 AM, "Kevin P. Fleming" wrote: On 09/15/2010 07:20 AM, Flavio Miranda wrote: > Recently I have instaled one Digium TDM410 on my... Right, that's what the message is telling you. For incoming calls on FXO, they can *only* be sent to the 's' extension in the target context, since there is no target number passed over the FXO connection. You'll have to create an 's' extension to handle incoming calls however you like. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] incoming call FXO
On 09/15/2010 07:20 AM, Flavio Miranda wrote: > Recently I have instaled one Digium TDM410 on my Asterisk. After > instaled , I can do outgoing calls but I cant receive calls. I receive > the following messages: > > chan_dahdi.c: Got event 2 (Ring/Answered)... > [Sep 14 11:24:44] NOTICE[2654] chan_dahdi.c: Got event 18 (Ring Begin)... > [Sep 14 11:24:44] WARNING[2654] pbx.c: Channel 'DAHDI/4-1' sent into > invalid extension 's' in context 'default', but no invalid handler > > I have not this 's' extension. Right, that's what the message is telling you. For incoming calls on FXO, they can *only* be sent to the 's' extension in the target context, since there is no target number passed over the FXO connection. You'll have to create an 's' extension to handle incoming calls however you like. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] incoming call FXO
Hi all, Recently I have instaled one Digium TDM410 on my Asterisk. After instaled , I can do outgoing calls but I cant receive calls. I receive the following messages: chan_dahdi.c: Got event 2 (Ring/Answered)...[Sep 14 11:24:44] NOTICE[2654] chan_dahdi.c: Got event 18 (Ring Begin)...[Sep 14 11:24:44] WARNING[2654] pbx.c: Channel 'DAHDI/4-1' sent into invalid extension 's' in context 'default', but no invalid handler I have not this 's' extension. Anybody knows what happen? Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming SIP Calls dumped to non-existent VM no matter what extensions.conf setup is used
On Wed, Aug 4, 2010 at 10:25 PM, Joe Wood wrote: > On Wed, Aug 4, 2010 at 7:49 PM, Warren Selby > wrote: > > On Wed, Aug 4, 2010 at 9:25 PM, Joe Wood wrote: > >> > > My experience with Asterisk in the past has been with inbound analog > lines so that would make sense :) > > See if you spot anything weird here: > > Try adding "insecure=invite" to the DID_NUMBER peer, reload SIP and try your call again. By the way, it looks like your SIP provider has a built-in auto-failover to voicemail setup. You may want to get them to disable that once you get everything working on your end. -- Thanks, --Warren Selby http://www.selbytech.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming SIP Calls dumped to non-existent VM no matter what extensions.conf setup is used
On Wed, Aug 4, 2010 at 7:49 PM, Warren Selby wrote: > On Wed, Aug 4, 2010 at 9:25 PM, Joe Wood wrote: >> >> I don't see any >> >> On Wed, Aug 4, 2010 at 7:04 PM, Warren Selby >> wrote: >> > >> > You don't have any extensions in your default context that match the >> > extension that your sip peer is dialing in on. 's' is not a default >> > extension for SIP...try using _X., and see what you get. Bump up the >> > CLI >> > (core set verbose 10) and then repost a failed called attempt. Some SIP >> > providers also use a + symbol in front of their inbound calls, so you >> > may >> > need to use _+X., instead. >> >> I don't see any call attempt/logs when I bump up the verbosity, and >> when I check my verbose logs I show: >> > > The next step would be to enable sip debug on the peer you're trying to > receive calls from (sip set debug peer PEERNAME or sip set debug ip > IPADDRESS). Then send another call inbound and see what's happening. As > far as the 's' extension, that's the "start" extension, it's used when no > other extension information is presented. Pretty much every SIP peer I've > ever seen presents an extension when entering a context, and thus the 's' > extension doesn't catch it. I've typically only seen 's' used in Macros and > with inbound analog lines. > My experience with Asterisk in the past has been with inbound analog lines so that would make sense :) See if you spot anything weird here: <--- SIP read from UDP:209.221.186.98:5060 ---> INVITE sip:s...@209.221.186.50 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 209.221.186.98;branch=z9hG4bK6b95.ae84c9620e19761029084a0f85255c29.0 Via: SIP/2.0/UDP 209.221.186.98:5071;branch=z9hG4bKf0a046579186bf058ba7783ed98a5a66;rport=5071 Max-Forwards: 16 From: 2538544199 ;tag=f7093e2d7e16a927d0816f6f5ed7aba4 To: Call-ID: ba711218-1ae3-122e-41bd-18a905721e58-b2b_1 CSeq: 200 INVITE Contact: Anonymous Expires: 300 User-Agent: Sippy Softswitch v2.0.80 cisco-GUID: 1225641884-3786690633-966044271-4144140181 h323-conf-id: 1225641884-3786690633-966044271-4144140181 Content-Length: 321 Content-Type: application/sdp v=0 o=- 1280279699622 1280279699622 IN IP4 209.221.186.98 s=- c=IN IP4 209.221.186.98 t=0 0 m=audio 60304 RTP/AVP 0 a=fmtp:4 bitrate=6300;annexa=no a=rtpmap:96 iLBC/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=20 a=fmtp:18 annexb=no a=rtpmap:98 telephone-event/8000 a=oldmediaip:208.76.155.20 a=nortpproxy:yes <-> [Aug 4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request: Header 0 [ 35]: INVITE sip:s...@209.221.186.50 SIP/2.0 [Aug 4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request: Header 1 [ 75]: Record-Route: [Aug 4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request: Header 2 [ 85]: Via: SIP/2.0/UDP 209.221.186.98;branch=z9hG4bK6b95.ae84c9620e19761029084a0f85255c29.0 [Aug 4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request: Header 3 [ 94]: Via: SIP/2.0/UDP 209.221.186.98:5071;branch=z9hG4bKf0a046579186bf058ba7783ed98a5a66;rport=5071 [Aug 4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request: Header 4 [ 16]: Max-Forwards: 16 [Aug 4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request: Header 5 [ 85]: From: 2538544199 ;tag=f7093e2d7e16a927d0816f6f5ed7aba4 [Aug 4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request: Header 6 [ 35]: To: [Aug 4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request: Header 7 [ 51]: Call-ID: ba711218-1ae3-122e-41bd-18a905721e58-b2b_1 [Aug 4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request: Header 8 [ 16]: CSeq: 200 INVITE [Aug 4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request: Header 9 [ 55]: Contact: Anonymous [Aug 4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request: Header 10 [ 12]: Expires: 300 [Aug 4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request: Header 11 [ 36]: User-Agent: Sippy Softswitch v2.0.80 [Aug 4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request: Header 12 [ 54]: cisco-GUID: 1225641884-3786690633-966044271-4144140181 [Aug 4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request: Header 13 [ 56]: h323-conf-id: 1225641884-3786690633-966044271-4144140181 [Aug 4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request: Header 14 [ 19]: Content-Length: 321 [Aug 4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request: Header 15 [ 29]: Content-Type: application/sdp [Aug 4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request: Header 16 [ 0]: [Aug 4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request:Body 0 [ 3]: v=0 [Aug 4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request:Body 1 [ 53]: o=- 1280279699622 1280279699622 IN IP4 209.221.186.98 [Aug 4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request:Body 2 [ 3]: s=- [Aug 4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request:Body 3 [ 23]: c=IN IP4 209.221.186.98 [Aug 4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request:Body 4 [ 5]: t=0 0 [Aug 4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request:Body 5 [ 23]: m=aud
Re: [asterisk-users] Incoming SIP Calls dumped to non-existent VM no matter what extensions.conf setup is used
On Wed, Aug 4, 2010 at 9:25 PM, Joe Wood wrote: > I don't see any > > On Wed, Aug 4, 2010 at 7:04 PM, Warren Selby > wrote: > > > > You don't have any extensions in your default context that match the > > extension that your sip peer is dialing in on. 's' is not a default > > extension for SIP...try using _X., and see what you get. Bump up the CLI > > (core set verbose 10) and then repost a failed called attempt. Some SIP > > providers also use a + symbol in front of their inbound calls, so you may > > need to use _+X., instead. > > I don't see any call attempt/logs when I bump up the verbosity, and > when I check my verbose logs I show: > > The next step would be to enable sip debug on the peer you're trying to receive calls from (sip set debug peer PEERNAME or sip set debug ip IPADDRESS). Then send another call inbound and see what's happening. As far as the 's' extension, that's the "start" extension, it's used when no other extension information is presented. Pretty much every SIP peer I've ever seen presents an extension when entering a context, and thus the 's' extension doesn't catch it. I've typically only seen 's' used in Macros and with inbound analog lines. -- Thanks, --Warren Selby http://www.selbytech.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming SIP Calls dumped to non-existent VM no matter what extensions.conf setup is used
I don't see any On Wed, Aug 4, 2010 at 7:04 PM, Warren Selby wrote: > > You don't have any extensions in your default context that match the > extension that your sip peer is dialing in on. 's' is not a default > extension for SIP...try using _X., and see what you get. Bump up the CLI > (core set verbose 10) and then repost a failed called attempt. Some SIP > providers also use a + symbol in front of their inbound calls, so you may > need to use _+X., instead. I don't see any call attempt/logs when I bump up the verbosity, and when I check my verbose logs I show: [Aug 4 19:16:42] VERBOSE[12287] pbx.c: -- Registered extension context 'default' (0xb77980c0) in local table 0xb77960c0; registrar: pbx_config [Aug 4 19:16:42] VERBOSE[12287] pbx.c: -- Added extension '_X.' priority 1 to default (0xb77980c0) [Aug 4 19:16:42] VERBOSE[12287] pbx.c: -- Added extension '_X.' priority 2 to default (0xb77980c0) [Aug 4 19:16:42] VERBOSE[12287] pbx.c: -- Added extension '_X.' priority 3 to default (0xb77980c0) [Aug 4 19:16:42] VERBOSE[12287] pbx.c: -- Added extension '_X.' priority 4 to default (0xb77980c0) [Aug 4 19:16:42] VERBOSE[12287] pbx.c: -- Added extension '_X.' priority 5 to default (0xb77980c0) [Aug 4 19:16:42] VERBOSE[12287] pbx.c: -- Added extension '_X.' priority 6 to default (0xb77980c0) [Aug 4 19:16:42] VERBOSE[12287] pbx.c: -- Added extension '_X.' priority 7 to default (0xb77980c0) [Aug 4 19:16:42] VERBOSE[12287] pbx.c: -- Added extension '_X.' priority 8 to default (0xb77980c0) [Aug 4 19:16:42] VERBOSE[12287] pbx.c: -- Registered extension context 'parkedcalls' (0xb7797ee0) in local table 0xb77960c0; registrar: features [Aug 4 19:16:42] VERBOSE[12287] pbx.c: -- merging incls/swits/igpats from old(parkedcalls) to new(parkedcalls) context, registrar = pbx_config [Aug 4 19:16:42] VERBOSE[12287] pbx.c: -- Added extension '700' priority 1 to parkedcalls (0xb7797ee0) [Aug 4 19:16:42] VERBOSE[12287] pbx.c: -- Time to scan old dialplan and merge leftovers back into the new: 0.89 sec [Aug 4 19:16:42] VERBOSE[12287] pbx.c: -- Time to restore hints and swap in new dialplan: 0.02 sec [Aug 4 19:16:42] VERBOSE[12287] pbx.c: -- Time to delete the old dialplan: 0.11 sec [Aug 4 19:16:42] VERBOSE[12287] pbx.c: -- Total time merge_contexts_delete: 0.000102 sec [Aug 4 19:17:04] VERBOSE[12255] netsock.c: == Using SIP RTP CoS mark 5 [Aug 4 19:19:04] VERBOSE[12255] netsock.c: == Using SIP RTP CoS mark 5 [Aug 4 19:21:39] VERBOSE[12255] netsock.c: == Using SIP RTP CoS mark 5 I get the same error. Same random voicemail when no voicemail is configured. I was under the impressing that "s" was the catchall for all incoming trunks. What has changed? Joe -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming SIP Calls dumped to non-existent VM no matter what extensions.conf setup is used
On Wed, Aug 4, 2010 at 8:52 PM, Joe Wood wrote: > Hello. > > I have been beating my head over this problem for about 6 hours now. > > I have a SIP peer, who I register to (successfully), who should be > directing all incoming calls at my [default] stanza in my > extensions.conf: > > [ Context 'default' created by 'pbx_config' ] > 's' =>1. Wait(1) > [pbx_config] >2. Answer() > [pbx_config] >3. Background(welcome) > [pbx_config] >4. Background(and) > [pbx_config] >5. Background(thank-you-for-calling) > [pbx_config] >6. Background(conference-reservations) > [pbx_config] >7. Waitfor() > [pbx_config] >8. Hangup() > [pbx_config] > > Unfortunately, no matter how I configure extensions.conf or sip.conf, > the phone call always ends up saying: "Extension is unavailable. > Please leave your message after the tone". > > sip.conf: > > [general] > register => NPANXX:passw...@service_provider_ip > registertimeout=29 > registerattempts=0 > defaultexpiry=60 > > [DID_NUMBER] > type=peer > context=default > host=SERVICE_PROVIDER_IP > authuser=DID_NUMBER > fromuser=DID_NUMBER > fromdomain=SERVICE_PROVIDER_REALM > remotesecret=SERVICE_PROVIDER_PASSWD > secret=SERVICE_PROVIDER_PASSWD > dtmfmode=rfc2833 > disallow=all > allow=ulaw > qualify=yes > > I am attempting just to get the starting point where I can direct > users through my asterisk box, but it won't direct users to the 's' > extention, only to some voicemail box. I've removed the voicemail > config. > > My extensions.conf is tiny: > > [globals] > > [general] > > [default] > exten => s,1,Wait(1) > exten => s,n,Answer() > exten => s,n,Background(welcome) > exten => s,n,Background(and) > exten => s,n,Background(thank-you-for-calling) > exten => s,n,Background(conference-reservations) > exten => s,n,Waitfor() > exten => s,n,Hangup() > > > What am I doing wrong here? > > > > Thanks for any help you can give. > > > Joe > You don't have any extensions in your default context that match the extension that your sip peer is dialing in on. 's' is not a default extension for SIP...try using _X., and see what you get. Bump up the CLI (core set verbose 10) and then repost a failed called attempt. Some SIP providers also use a + symbol in front of their inbound calls, so you may need to use _+X., instead. -- Thanks, --Warren Selby http://www.selbytech.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Incoming SIP Calls dumped to non-existent VM no matter what extensions.conf setup is used
Hello. I have been beating my head over this problem for about 6 hours now. I have a SIP peer, who I register to (successfully), who should be directing all incoming calls at my [default] stanza in my extensions.conf: [ Context 'default' created by 'pbx_config' ] 's' =>1. Wait(1)[pbx_config] 2. Answer() [pbx_config] 3. Background(welcome)[pbx_config] 4. Background(and)[pbx_config] 5. Background(thank-you-for-calling) [pbx_config] 6. Background(conference-reservations)[pbx_config] 7. Waitfor() [pbx_config] 8. Hangup() [pbx_config] Unfortunately, no matter how I configure extensions.conf or sip.conf, the phone call always ends up saying: "Extension is unavailable. Please leave your message after the tone". sip.conf: [general] register => NPANXX:passw...@service_provider_ip registertimeout=29 registerattempts=0 defaultexpiry=60 [DID_NUMBER] type=peer context=default host=SERVICE_PROVIDER_IP authuser=DID_NUMBER fromuser=DID_NUMBER fromdomain=SERVICE_PROVIDER_REALM remotesecret=SERVICE_PROVIDER_PASSWD secret=SERVICE_PROVIDER_PASSWD dtmfmode=rfc2833 disallow=all allow=ulaw qualify=yes I am attempting just to get the starting point where I can direct users through my asterisk box, but it won't direct users to the 's' extention, only to some voicemail box. I've removed the voicemail config. My extensions.conf is tiny: [globals] [general] [default] exten => s,1,Wait(1) exten => s,n,Answer() exten => s,n,Background(welcome) exten => s,n,Background(and) exten => s,n,Background(thank-you-for-calling) exten => s,n,Background(conference-reservations) exten => s,n,Waitfor() exten => s,n,Hangup() What am I doing wrong here? Thanks for any help you can give. Joe -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming call doesn't finish when internal phone hangs up
Hi Elder, I would first check the behaviour of your PSTN lines (i.e. nothing to do with Asterisk). In many places PSTN companies allow between 30 to 90 seconds of connection to remain open even if the -called- party, NOT the calling party, has hung-up. Normally this is to allow putting down the phone in one room and picking up in another room without disconnecting the line. Make a small test to verify this and if this is the case you will need to discuss this with your PSTN provider. Harel Date: Thu, 8 Jul 2010 12:01:40 -0500 From: Daniel - Asterisk Subject: [asterisk-users] Incoming call doesn't finish when internal phone hangs up To: Asterisk Users Mailing List - Non-Commercial Discussion Message-ID: Content-Type: text/plain; charset=ISO-8859-1 Hello guys, I have this problem when a call is received in my PBX: (Caller) --> (Redirecting Service) --> (E1 PRI) --> (Asterisk PBX) --> (Internal Phone) Reception works fine, but when conversation finishes and the agent at internal phone hangs up, the call at caller's side is still alive for many seconds until it hangs up. The problem is that Telephone Company is billing me acording Caller's duration which is bigger than Asterisk's CDR. The same issue happens when Caller dials E1 PRI directly. In every case Asterisk finishes normally the call as CDR and CLI register correctly. I'm using Asterisk 1.4.21.2 and OpenVox DE210P card. Configuration files follow: zaptel.conf: span=1,1,1,ccs,hdb3 bchan=1-15,17-31 dchan=16 # Span 2: TE4/0/2 "T4XXP (PCI) Card 0 Span 2" span=2,2,1,ccs,hdb3 bchan=32-46,48-62 dchan=47 # Global data loadzone= us defaultzone = us zapata.conf: [channels] language=es context=default rxwink=300 usecallerid=yes hidecallerid=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 busydetect=yes busycount=yes busypattern=500,500 answeronpolarityswitch=yes hanguponpolarityswitch=yes ;PRI RDSI - SPAN 1 group = 1 context = incoming-1 inmediate=no switchtype=euroisdn signalling=pri_cpe channel => 1-15,17-31 ;PRI RDSI - SPAN 2 group = 1 context = incoming-2 inmediate=no switchtype=euroisdn signalling=pri_cpe channel => 32-46,48-62 ... Thanks in advance, Elder Arohuanca Lagos Phone: +51 1 991696900 Lima - Peru -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users