[asterisk-users] Asterisk Sip trunking routing problem
Hi This problem is driving me crazy. I have two severs which are trunked by the sip Asterisk box A is natted thus in my sip.conf file I use the following externip=DDNS address Asterisk box B is NOT natted and has a static IP. Asterisk box A and B are both registered with each other. B holds the address of the DDNS public IP of A for the sip trunk. Everything works fine. The problem comes in when on BOX A I make a seperate PPPOE connection to route only VOIP packets to BOX B. BOX B still gets the DDNS address of BOX A and not the new PPPOE address This causes all sip requests to be rejected on BOX B due to authenication failer I traced it down the problem to the externip command in sip.conf When I turn off externip, BOX B gets the correct PPPOE IP of BOX A. I dont think I can just disable externip due to natting as other sip clients need to access BOX A outside of its network. Anyone have any ideas on this problem or how to solve it Zakir-- _ -- 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] Sip Hangup after critical packet
HI I have asterisk 1.6.13 running. My ATA are connected via vpn (openvpn) Out going calls from asterisk to the ata works fine Incoming calls from the ata to asterisk cuts off with the error msg Maximum retries exceeded on transmission 70854efe-4157e...@10.168.7.103 for seqno 102 (Critical Response) -- See doc/sip-retransmit.txt. [Dec 6 13:52:43] WARNING[3921]: chan_sip.c:3858 retrans_pkt: Hanging up call 70854efe-4157e...@10.168.7.103 - no reply to our critical packet (see doc/sip-retransmit.txt). I been googling this error and it was mentioned to use t1min= 500 however its only delaying the problem. any ideas on what is the cause of this problem. Only 2-3 atas are having this problem the rest are fine zakir-- _ -- 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] Fw: Sip Hangup after critical packet SIP DEBUG attached
HI I have asterisk 1.6.13 running. My ATA are connected via vpn (openvpn) Out going calls from asterisk to the ata works fine Incoming calls from the ata to asterisk cuts off with the error msg Maximum retries exceeded on transmission 70854efe-4157e...@10.168.7.103 for seqno 102 (Critical Response) -- See doc/sip-retransmit.txt. [Dec 6 13:52:43] WARNING[3921]: chan_sip.c:3858 retrans_pkt: Hanging up call 70854efe-4157e...@10.168.7.103 - no reply to our critical packet (see doc/sip-retransmit.txt). I been googling this error and it was mentioned to use t1min= 500 however its only delaying the problem. any ideas on what is the cause of this problem. Only 2-3 atas are having this problem the rest are fine. Here is the sip debug the sip invites are not being received and in one of the message a busy response was sent back. Retransmitting #4 (no NAT) to 10.168.7.103:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.168.7.103:5060;branch=z9hG4bK-60ef482f;received=10.168.7.103 From: Ridge ;tag=f314fd35733eba9bo0 To: ;tag=as4593172b Call-ID: f54a1cbd-891ce...@10.168.7.103 CSeq: 102 INVITE Server: Asterisk PBX 1.6.2.13 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 337 v=0 o=root 777980638 777980638 IN IP4 41.146.208.131 s=Asterisk PBX 1.6.2.13 c=IN IP4 41.146.208.131 t=0 0 m=audio 19726 RTP/AVP 8 0 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- Retransmitting #5 (no NAT) to 10.168.7.103:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.168.7.103:5060;branch=z9hG4bK-60ef482f;received=10.168.7.103 From: Ridge ;tag=f314fd35733eba9bo0 To: ;tag=as4593172b Call-ID: f54a1cbd-891ce...@10.168.7.103 CSeq: 102 INVITE Server: Asterisk PBX 1.6.2.13 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 337 v=0 o=root 777980638 777980638 IN IP4 41.146.208.131 s=Asterisk PBX 1.6.2.13 c=IN IP4 41.146.208.131 t=0 0 m=audio 19726 RTP/AVP 8 0 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- Reliably Transmitting (no NAT) to 10.168.7.103:5060: OPTIONS sip:2...@10.168.7.103:5060 SIP/2.0 Via: SIP/2.0/UDP 41.146.208.131:5060;branch=z9hG4bK226c4b89;rport Max-Forwards: 70 From: "asterisk" ;tag=as21bdce7e To: Contact: Call-ID: 062f8f6c4e7f5929487f3db12a93f...@41.146.208.131 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.2.13 Date: Mon, 06 Dec 2010 12:47:41 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- <--- SIP read from UDP:10.168.7.103:5060 ---> SIP/2.0 486 Busy Here To: ;tag=18c8b9ab85ca5068i0 From: "asterisk" ;tag=as21bdce7e Call-ID: 062f8f6c4e7f5929487f3db12a93f...@41.146.208.131 CSeq: 102 OPTIONS Via: SIP/2.0/UDP 41.146.208.131:5060;branch=z9hG4bK226c4b89 Server: Linksys/SPA3102-3.3.6(GW) Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura <-> --- (10 headers 0 lines) --- Really destroying SIP dialog '062f8f6c4e7f5929487f3db12a93f...@41.146.208.131' Method: OPTIONS Retransmitting #6 (no NAT) to 10.168.7.103:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.168.7.103:5060;branch=z9hG4bK-60ef482f;received=10.168.7.103 From: Ridge ;tag=f314fd35733eba9bo0 To: ;tag=as4593172b Call-ID: f54a1cbd-891ce...@10.168.7.103 CSeq: 102 INVITE Server: Asterisk PBX 1.6.2.13 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 337 v=0 o=root 777980638 777980638 IN IP4 41.146.208.131 s=Asterisk PBX 1.6.2.13 c=IN IP4 41.146.208.131 t=0 0 m=audio 19726 RTP/AVP 8 0 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv zakir-- _ -- 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] Inter pbx communication via BRI
hi I am setting up a test lab using BRI to connect two pbxs together. I am new to BRI / ISDN. For now I am just need some advice on what hardware which supports NT mode to use. I am also looking for info on setting up the BRI card in asterisks, setting the channels, mode type etc. Once I got a good grip on ISDN BRI basic rate, my goal is to have 2 BRI ports which will be connected to Telco and 2 Ports for Inter PBX. The hardware should be able to support NT/TE modes configuration for each port. thanks in advance Zakir -- _ -- 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] session border controller
Hi I am playing around the idea of setting up an asterisk box on the public domain. This box will then be connected to various other sip providers for LCR for low cost calls. It was recommened that I use a SBC. I never used SBC. From what I understand it can help with different aspect when it comes to SIP communication such as security and billing. Is SBC really necessary? If so what would be the ideal configuration? thanks Zakir-- _ -- 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] SIP 401
Hi I am trying to get 2 accounts from voipblaster to talk to each other. Calls withing voipblaster network is free. If I configure two sip clients with the two accounts it works fine however with Asterisk I am getting SIP 401 In my Sip.conf file I under general register = user:passw...@sip.voipblaster.com then I have a sip peer [FreeCall](default) type= friend context= incoming username = kiks2010 secret = password host= sip.voipblast.com fromuser = kiks2010 fromdomain = sip.voipblast.com insecure=very qualify=yes these are the sip debug logs v=0 o=kiks2010 1287592622 1287592622 IN IP4 77.72.168.99 s=SIP Call c=IN IP4 77.72.168.99 t=0 0 m=audio 11538 RTP/AVP 8 101<-> --- (11 headers 9 lines) --- == Using SIP RTP CoS mark 5 Sending to 77.72.174.128 : 5060 (NAT) Using INVITE request as basis request - 64de05c42e7b4ef2a0678f999c0ed...@77.72.174.128 Found peer 'FreeCall' for 'ajs2010' from 77.72.174.128:5060 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=ptime:20 <--- Reliably Transmitting (NAT) to 77.72.174.128:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 77.72.174.128:5060;branch=z9hG4bK6ff0e241f3fd4d0b9c137d616de1fe1f;received=77.72.174.128 From: "ajs2010" ;tag=330113ac4c51ef02d4ef70 Any help info will be appreciated thanks Zakir-- _ -- 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] 1 way audio asterisk 1.6
Hi I wonder if anyone could give some light on SIP NAT. I've having a friken headache with SIP NAT 1 way audio. Client - NAT - NAT - Server Client can hear users from server side but server cant hear client. Ive tried every possible settings externip set localip set NAT= yes / route directmedia yes/ no Ive check the sip headers in the debug mode and its using the external address in both client and server. Ive tried STUn servers etc No luck. any info on this Its for my installation which I am testing. Zakir-- _ -- 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] mISDN issues again
Hi I been testing my clients installation for any problems with the farsouth gateway and what I noticed is that the calls are being dropped at transfer. After testing I found that when the reception hits the transfer button on her sip phone, the caller gets a dialtone and the reception goes on hold. Very wierd problem. Ive disabled senddtmf and astdtmf in misdn.conf , that seems to have solved the problem. Any insight on this. Zakir-- _ -- 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] gigasets A580IP Recall Button
Hi I am trying to get the recall button working for the gigasets What settings do i need to set in the advance settings? Zakir-- _ -- 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] Billing
Hi I have the following topology: Sever A <> IAX2 <-> SERVER B <> SIP <> ITSP Server A : Branch Server ( with billing cdr ) : Billing module for auditing calls at branch level Server B : Billing Server ( with billing cdr ) : Official statements get generating from this server I am trying to match my billsecs on server A TO that of server B The problem seems to be that server A is counting the ringing as well, not just the ANSwered ( person talking ) If the phone rings for 25 secs and we speak for 30 secs, billsec is 55 secs and duration is 55 secs CDR values for deposition are correct if if the call has not been answered or busy. ie NO ANSWER or BUSY on Server A. Asterisk 13 on branch server any ideas on how to trouble shoot this problemthanks Zakir -- _ -- 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] Setup DID
Hi I am trying to setup DDI for one of our servers Our Provider has given us one DDI for use for eg 080011. On my main server A, I use an IAX trunk to connect to Client Server B.When calls come in from the outside world on main server A for 080011In the dial plan, I pattern match and connect with IAX2 truck named 087XX eg) SERVER A DIALPLANexten => 080011,1, Verbose( 3, " INCOMING CALLS SERVER B )same => n,Dial(IAX2/087XX,,r) On server B I have an incoming context in which I have both a general IVR and the 080011 patternto route the DDI number to a particular extension. [incoming] exten =>080011.,1,Answer()same => n,Dial(PJSIP/200)same => n,Hangup() exten => s,1,Answer()same => n,Goto(main_ivr,,1)same => n,Hangup() I cant seem to get a match on 087, it always go to the s context in incomingAny ideas on how I can get DDI to work thanks Zakir -- _ -- 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] asterisk callerid issue PJSIP Realtime
I recently rolled out a new server with asterisk 14. On the Called user phone,
the caller ID is the same as the Called User.
eg) ext 406 with callerid 406 calls ext 405 , on the caller id on the ext
405 phone displaying 405.
We are using realtime PJSIP, I set the callerid field in the database but no
luck.
- Executing [405@common:1] NoOp("PJSIP/406-000f", ""DEBUGGING PJSIP CLID"")
in new stack
- Executing [405@common:2] NoOp("PJSIP/406-000f", "CALLERID = "ross"
<406>") in new stack- Executing [405@common:3] Dial("PJSIP/406-000f",
"PJSIP/405") in new stack
In the above dialplan, the callerid is been taken from the database PJSIP
endpoints.
Here is the sip debugger files
INVITE sip:405@192.168.1.27 SIP/2.0Via: SIP/2.0/UDP
192.168.1.82:5060;branch=z9hG4bK714210067;rportFrom: "zak"
;tag=2071662084To: Call-ID:
50172054-506...@bjc.bgi.B.ICCSeq: 21 INVITEContact: "zak"
Authorization: Digest username="406",
realm="asterisk", nonce="1485956409/e852b2a5e081f01421212d9a6ca954fa",
uri="sip:405@192.168.1.27", response="ef94bae123f16dc5d9314a43922c949d",
algorithm=md5, cnonce="13226017", opaque="50d490d233efd03e", qop=auth,
nc=0003
INVITE sip:405@192.168.1.209:36767;ob SIP/2.0Via: SIP/2.0/UDP
197.245.99.113:5060;rport;branch=z9hG4bKPj2f9d3dde-5ec4-49e1-b92d-7b4091b3138bFrom:
;tag=e4a0ecf6-c74e-4ab5-8438-bac5c073e328To:
Contact: Call-ID:
b4a83465-9105-4c70-9da1-11f410c37657
<--- Received SIP response (515 bytes) from UDP:192.168.1.209:36767 --->SIP/2.0
180 RingingVia: SIP/2.0/UDP
197.245.99.113:5060;rport=5060;received=192.168.1.27;branch=z9hG4bKPj70fb8ef9-d99c-4e5b-88a5-eecbf7dd7682Call-ID:
f0b31a86-0ac3-47f0-8b13-487d54982e9bFrom:
;tag=77ea8869-273a-4f65-8128-e334b445f970To:
;tag=jurMewPN-95CgNyoQbhRCFpbH90hKw1dCSeq: 12221
INVITEContact: Allow: PRACK, INVITE, ACK, B
ParameterName : ParameterValue
= callerid
: "john doe" <405> callerid_privacy : allowed
callerid_tag :
Zakir
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Re: [asterisk-users] asterisk callerid issue PJSIP Realtime
Yes, from_user was set, removing those entries solved the problem.
Can someone please explain to me the correct use for fromuser field?
thanksZakir
On Wednesday, February 1, 2017 8:00 PM,
"asterisk-users-requ...@lists.digium.com"
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Today's Topics:
1. asterisk callerid issue PJSIP Realtime (Zakir Mahomedy)
2. Re: asterisk callerid issue PJSIP Realtime (George Joseph)
--
Message: 1
Date: Wed, 1 Feb 2017 13:50:57 + (UTC)
From: Zakir Mahomedy
To: "asterisk-users@lists.digium.com"
Subject: [asterisk-users] asterisk callerid issue PJSIP Realtime
Message-ID: <1998594554.250932.1485957057...@mail.yahoo.com>
Content-Type: text/plain; charset="utf-8"
I recently rolled out a new server with asterisk 14. ?On the Called user phone,
the caller ID is the same as the Called User.
eg) ext ?406 ?with callerid 406 ? calls ext 405 ,??on the caller id on the ext
405 phone displaying 405.
We are using realtime PJSIP, I set the callerid field in the database but no
luck.?
- Executing [405@common:1] NoOp("PJSIP/406-000f", ""DEBUGGING PJSIP CLID"")
in new stack
- Executing [405@common:2] NoOp("PJSIP/406-000f", "CALLERID = ?"ross"
<406>") in new stack- Executing [405@common:3] Dial("PJSIP/406-000f",
"PJSIP/405") in new stack
In the above dialplan, the callerid is been taken from the database PJSIP
endpoints.?
Here is the sip debugger files
INVITE sip:405@192.168.1.27 SIP/2.0Via: SIP/2.0/UDP
192.168.1.82:5060;branch=z9hG4bK714210067;rportFrom: "zak"
;tag=2071662084To: Call-ID:
50172054-506...@bjc.bgi.B.ICCSeq: 21 INVITEContact: "zak"
Authorization: Digest username="406",
realm="asterisk", nonce="1485956409/e852b2a5e081f01421212d9a6ca954fa",
uri="sip:405@192.168.1.27", response="ef94bae123f16dc5d9314a43922c949d",
algorithm=md5, cnonce="13226017", opaque="50d490d233efd03e", qop=auth,
nc=0003
INVITE sip:405@192.168.1.209:36767;ob SIP/2.0Via: SIP/2.0/UDP
197.245.99.113:5060;rport;branch=z9hG4bKPj2f9d3dde-5ec4-49e1-b92d-7b4091b3138bFrom:
;tag=e4a0ecf6-c74e-4ab5-8438-bac5c073e328To:
Contact: Call-ID:
b4a83465-9105-4c70-9da1-11f410c37657
<--- Received SIP response (515 bytes) from UDP:192.168.1.209:36767 --->SIP/2.0
180 RingingVia: SIP/2.0/UDP
197.245.99.113:5060;rport=5060;received=192.168.1.27;branch=z9hG4bKPj70fb8ef9-d99c-4e5b-88a5-eecbf7dd7682Call-ID:
f0b31a86-0ac3-47f0-8b13-487d54982e9bFrom:
;tag=77ea8869-273a-4f65-8128-e334b445f970To:
;tag=jurMewPN-95CgNyoQbhRCFpbH90hKw1dCSeq: 12221
INVITEContact: Allow: PRACK, INVITE, ACK, B
?ParameterName ? ? ? ? ? ? ? ? ? ? ?:
ParameterValue?=?callerid
? ? ? ? ? ? ? ? ? ? ? ? ? : "john doe" <405>?callerid_privacy ? ? ? ? ? ? :
allowed?callerid_tag ? ? ? ? ? ? ? ? ? ?:
Zakir
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Message: 2
Date: Wed, 1 Feb 2017 08:52:59 -0700
From: George Joseph
To: Zakir Mahomedy , Asterisk Users Mailing List
- Non-Commercial Discussion
Subject: Re: [asterisk-users] asterisk callerid issue PJSIP Realtime
Message-ID:
Content-Type: text/plain; charset="utf-8"
On Wed, Feb 1, 2017 at 6:50 AM, Zakir Mahomedy wrote:
> I recently rolled out a new server with asterisk 14.
> On the Called user phone, the caller ID is the same as the Called User.
>
> eg) ext 406 with callerid 406 calls ext 405 ,
>
> on the caller id on the ext 405 phone displaying 405.
>
>
>
> We are using realtime PJSIP, I set the callerid field in the database but
> no luck.
>
> - Executing [405@common:1] NoOp("PJSIP/406-000f", ""DEBUGGING PJSIP
> CLID"") in new stack
> - Executing [405@common:2] NoOp("PJSIP/406-000f", "CALLERID = "ross"
> <406>") in new stack
> - Executing [405@common:3] Dial("PJSIP/406-000f", "PJSIP/405") in new
> stack
>
> In the above dialplan, the callerid is been
