[asterisk-users] Can ChanIsAvail return status from sip uri using router ip
hello, Although my previous posts in this forum have not received satisfying answers, here is another question from me. my question is can i use ChanIsAvail function to get the status of a user in the format SPI/user-id if i provide user in sip uri like this ChanIsAvail(SIP/u...@153.18.x.x:5062) calling user with this sip uri works fine. I once tried but status returned was unknow host 153.18.x.x. what is wrong here? thanks Nasir Javaid -- _ -- 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] RTP stream not passing through router with port forwarding
Hi, I am trying to dial a registered user via his IP:Port mechanism, but problem is that the audio data is not reaching to dialed user. here is the scenario. caller and callee both are registered at asterisk server. asterisk server is on public ip so no port forwarding and natting necessary there. however caller and callee both are behind router and there is port forwarding enabled and nat=yes, qualify=yes in sip.conf for both users. callee user name:adf callee local ip/port: 192.168.0.10:5678 callee router ip: 116.79.x.x when we simply dial callee as Dial(SIP/adf) RTP stream reaches perfectly fine to 192.168.0.10 through router and INVITE is sent to local ip through router. INVITE sip:a...@192.168.0.10:5678 SIP/2.0 (asterisk somehow manages to contact local ip through router and sends rtp there) but problem arises when i dial using IP:Port combination like this Dial(SIP/a...@116.79.x.x:5678) In this case INVITE is sent to router ip instead of local ip through router. INVITE sip:a...@116.79.x.x:5678 SIP/2.0 (asterisk sends rtp to router ip and not local ip) Similerly TO header also has same ip as INVITE. I think in IP dial rtp is not reaching to local ip through router as INVTE is meant for router ip and asterisk does not know where to send rtp stream after sending it to router. how can this issue be resolved? is there something to be done at router confiurations or sip.conf parameters. I have already played with nat/qualify/canreinvite/directrtp/externip etc parameters. regards, Nasir Javaid -- _ -- 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] asterisk-users Digest, Vol 73, Issue 5
Hi C F
no asterisk and sip device are not behind same router. actually both are in
different countries. how ever when caller and callee are behind same routers
voice is just fine (both ways) and i can see re-INVITEs too.
but when someone calls from another router then this issue arises. caller
can hear the called party but called party can not hear caller. and there
are no re-invites issued too.
i am bit new to sip and rtp stuff and don't know what is going on. how
asterisk is issuing re-invites for devices behind same router and not for
device behind another router?
Nasir Javaid
Message: 12
Date: Tue, 3 Aug 2010 07:21:06 -0400
From: C F shma...@gmail.com
Subject: Re: [asterisk-users] RTP stream not passing through router
withport forwarding
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID:
aanlktin9g14ipfl3yvmsfqmtiy=b9wgfci4xerdrb...@mail.gmail.com
Content-Type: text/plain; charset=ISO-8859-1
Is asterisk and the SIP device behind the same router?
Most routers will not redirect internal NAT requests. So that if you
are trying to have port forwarding done but the request and the
forwarding destination are on the same interface it won't work.
On 8/3/10, Nasir Javaid nasirjavaidna...@gmail.com wrote:
Hi,
I am trying to dial a registered user via his IP:Port mechanism, but
problem
is that the audio data is not reaching to dialed user. here is the
scenario.
caller and callee both are registered at asterisk server. asterisk server
is
on public ip so no port forwarding and natting necessary there. however
caller and callee both are behind router and there is port forwarding
enabled and nat=yes, qualify=yes in sip.conf for both users.
callee user name:adf
callee local ip/port: 192.168.0.10:5678
callee router ip: 116.79.x.x
when we simply dial callee as Dial(SIP/adf) RTP stream reaches perfectly
fine to 192.168.0.10 through router and INVITE is sent to local ip
through
router.
INVITE sip:a...@192.168.0.10:5678 SIP/2.0 (asterisk somehow manages to
contact local ip through router and sends rtp there)
but problem arises when i dial using IP:Port combination like this
Dial(SIP/a...@116.79.x.x:5678)
In this case INVITE is sent to router ip instead of local ip through
router.
INVITE sip:a...@116.79.x.x:5678 SIP/2.0 (asterisk sends rtp to router
ip
and not local ip)
Similerly TO header also has same ip as INVITE. I think in IP dial rtp is
not reaching to local ip through router as INVTE is meant for router ip
and
asterisk does not know where to send rtp stream after sending it to
router.
how can this issue be resolved? is there something to be done at router
confiurations or sip.conf parameters. I have already played with
nat/qualify/canreinvite/directrtp/externip etc parameters.
regards,
Nasir Javaid
--
Message: 13
Date: Tue, 03 Aug 2010 13:21:23 +0200
From: Philipp von Klitzing klitz...@pool.informatik.rwth-aachen.de
Subject: Re: [asterisk-users] mapping of disconnect reasons
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID:
4c5817d3.976.37d9...@klitzing.pool.informatik.rwth-aachen.de
Content-Type: text/plain; charset=ISO-8859-1
Hi!
Is there a way to change the mappings of disconnect reasons to certain
SIP messages? E.G. I need to change the mapping for SIP 402 Payment
Required from 16 (normal termination) like it is in 1.4.24 to 21
(call rejected) as defined in RFC 3398.
* if you think the mapping is wrong, then you should open a ticket on the
Asterisk bug tracker
* the mapping can only be changed in the code - which you ahve
* Asterisk 1.8 will allow to read SIP response codes in the dialplan via
{HASH(SIP_CAUSE,channel-name)}. Asterisk 1.8 also comes with a
'use_q850_reason' configuration option for generating and parsing, if
available, Reason: Q.850;cause=cause code.
Philipp
--
Message: 14
Date: Tue, 03 Aug 2010 13:21:23 +0200
From: Philipp von Klitzing klitz...@pool.informatik.rwth-aachen.de
Subject: Re: [asterisk-users] Codec negotiation : expecting G726,
getting G711a (alaw)
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID:
4c5817d3.31146.37d9...@klitzing.pool.informatik.rwth-aachen.de
Content-Type: text/plain; charset=US-ASCII
Hi!
Question 1 :
[Aug 2 13:56:57] Capabilities: us - 0x90a (gsm|alaw|g726|g729), peer -
audio=0x808 (alaw|g726)/video=0x0 (nothing), combined - 0x808 (alaw|g726)
why is combined alaw|g726 and not g726|alaw (reverse) ??
Guess: Here the order presented has no meaning for the order of codec
negotiation.
Question 2 :
why do I see on my Grandstream phone that the codec being used is alaw in
stead of g726
Re: [asterisk-users] asterisk-users Digest, Vol 72, Issue 81
thanks for your reply but i did not meant that. ${CALLERID(DNID)} will
return then number which i don't want. what i want is channel-id like if we
have a user named nasir, then we dial it as follows
Dial(SIP/nasir)
but actual channel-id that asterisk uses is something like nasir-2b487e9.
and on the asterisk cli we can check this when call is answered or hangup,
asterisk attaches some random id with username.
i am dialing sip uri using Dial(SIP/119.26.18.235:5062) which causes
changed INVITE adn TO headers, so i want to get the channel-id that asterisk
internally uses do dial it.
if we use ChanIsAvail(SIP/nasir) or ChanIsAvail(SIP/192.168.0.10:5062) this
works on Local LAN and it returns SIP/192.168.0.10:5062-3fe934f4 , but
when asterisk is on Live Ip and users are behind Router then this function
gives error of unknow host. so i want to know if there is any other function
that does this job.
so what is want is to get this channel-id ( like nasir-2487e9) and dial it
like
Dial(SIP/nasir-2487e9) or Dial(SIP/119.26.18.235:5062-34e984b)
hope this clears what i wanna do.
Message: 8
Date: Thu, 29 Jul 2010 10:37:07 -0500
From: Danny Nicholas da...@debsinc.com
Subject: Re: [asterisk-users] How to extract channel-id of a user or
peer
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Message-ID: 201007291515.o6tffv8t025...@mail.debsinc.com
Content-Type: text/plain; charset=us-ascii
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nasir Javaid
Subject: [asterisk-users] How to extract channel-id of a user or peer
my question is how can i get channel-id of a user or peer. I tried using
ChanIsAvail(username). this works correctly when user and asterisk are on
Local LAN. But my asterisk server is on public ip and users are behind nat,
and so this method says unknow host when used on public asterisk server.
I also tried built-in variable ${CHANNEL}, but this returns the channel-id
of the calling channel. but i want channel-id of called user.
--
perhaps ${CALLERID(DNID)}
--
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Re: [asterisk-users] asterisk-users Digest, Vol 72, Issue 82
thanks for your reply but i think ${BRIDGEPEER} will work only when both
channels are connected. i want to get channel-id before dialing so that i
can dial using that channel id.
${BRIDGEPEER} is probably a good way to do what you want.. if Channel
A calls Channel B, and you want Channel A to get the channelID of
Channel B, as long as the two channels are bridged, ${BRIDGEPEER} will
do what you want
perhaps ${CALLERID(DNID)}
my question is how can i get channel-id of a user or peer. I tried using
ChanIsAvail(username). this works correctly when user and asterisk are
on
Local LAN. But my asterisk server is on public ip and users are behind
nat,
and so this method says unknow host when used on public asterisk server.
I also tried built-in variable ${CHANNEL}, but this returns the
channel-id
of the calling channel. but i want channel-id of called user.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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[asterisk-users] Nat issue one way audio on IP dial
thanks Jim I will check stun server settings asap, but i have noticed 192.168.x.x is also present in the debug of successful call having both way audio. so i don't think this has to do anything with this. below is the sip debug of successful call . --- Audio is at 79.80.154.99 port 14034 Adding codec 0x8 (alaw) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 116.18.35.235:28614: INVITE sip:a...@116.18.35.235:28614;rinstance=0266b8b94f488588 SIP/2.0 Via: SIP/2.0/UDP 79.80.154.99:5678;branch=z9hG4bK5acbd17f;rport From: pepsi coke sip:12345678...@79.80.154.99:5678;tag=as12245807 To: sip:a...@116.18.35.235:28614;rinstance=0266b8b94f488588 Contact: sip:12345678...@79.80.154.99:5678 Call-ID: 25a6e3604896da0e5482a7565560c...@79.80.154.99 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 21 Jul 2010 15:06:24 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 285 v=0 o=root 9626 9626 IN IP4 79.80.154.99 s=session c=IN IP4 79.80.154.99 t=0 0 m=audio 14034 RTP/AVP 8 0 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Jul 21 11:06:24] WARNING[23749]: chan_sip.c:2872 sip_call: Setting auto-congest time to 15000 ms. -- Called adf ast-server*CLI --- SIP read from 116.18.35.235:28614 --- SIP/2.0 180 Ringing Via: SIP/2.0/UDP 79.80.154.99:5678;branch=z9hG4bK5acbd17f;rport=5678 Contact: sip:a...@116.18.35.235:28614;rinstance=0266b8b94f488588 To: sip:a...@116.18.35.235:28614;rinstance=0266b8b94f488588;tag=bd6f2350 From: pepsi cokesip:12345678...@79.80.154.99:5678;tag=as12245807 Call-ID: 25a6e3604896da0e5482a7565560c...@79.80.154.99 CSeq: 102 INVITE User-Agent: X-Lite release 1104o stamp 56125 Content-Length: 0 - --- (9 headers 0 lines) --- -- SIP/adf-00794e30 is ringing ast-server*CLI --- SIP read from 116.18.35.235:28614 --- SIP/2.0 200 OK Via: SIP/2.0/UDP 79.80.154.99:5678;branch=z9hG4bK5acbd17f;rport=5678 Contact: sip:a...@116.18.35.235:28614;rinstance=0266b8b94f488588 To: sip:a...@116.18.35.235:28614;rinstance=0266b8b94f488588;tag=bd6f2350 From: pepsi cokesip:12345678...@79.80.154.99:5678;tag=as12245807 Call-ID: 25a6e3604896da0e5482a7565560c...@79.80.154.99 CSeq: 102 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: X-Lite release 1104o stamp 56125 Content-Length: 185 v=0 o=- 2 2 IN IP4 192.168.0.12 s=CounterPath X-Lite 3.0 c=IN IP4 192.168.0.12 t=0 0 m=audio 15956 RTP/AVP 8 0 101 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendrecv - --- (11 headers 9 lines) --- Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 192.168.0.12:15956 Found description format telephone-event for ID 101 Capabilities: us - 0x10e (gsm|ulaw|alaw|g729), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.0.12:15956 list_route: hop: sip:a...@116.18.35.235:28614;rinstance=0266b8b94f488588 [Jul 21 11:06:38] DEBUG[9707]: chan_sip.c:5695 reqprep: Strict routing enforced for session 25a6e3604896da0e5482a7565560c...@79.80.154.99 set_destination: Parsing sip:a...@116.18.35.235:28614;rinstance=0266b8b94f488588 for address/port to send to set_destination: set destination to 116.18.35.235, port 28614 Transmitting (NAT) to 116.18.35.235:28614: ACK sip:a...@116.18.35.235:28614;rinstance=0266b8b94f488588 SIP/2.0 Via: SIP/2.0/UDP 79.80.154.99:5678;branch=z9hG4bK00fdcc7c;rport From: pepsi coke sip:12345678...@79.80.154.99:5678;tag=as12245807 To: sip:a...@116.18.35.235:28614;rinstance=0266b8b94f488588;tag=bd6f2350 Contact: sip:12345678...@79.80.154.99:5678 Call-ID: 25a6e3604896da0e5482a7565560c...@79.80.154.99 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- -- Call on SIP/adf-00794e30 left from hold -- SIP/adf-00794e30 answered SIP/pepsi-9fd06cc0 ast-server*CLI --- SIP read from 116.18.35.235:28614 --- - --- (0 headers 1 lines) --- ast-server*CLI --- SIP read from 116.18.35.235:28614 --- SUBSCRIBE sip:a...@ast-server.axvoice.com:5678 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.12:28614 ;branch=z9hG4bK-d8754z-7039d4338568107f-1---d8754z-;rport Max-Forwards: 70 Contact: sip:a...@116.18.35.235:28614 To: adfsip:a...@ast-server.axvoice.com:5678 From: adfsip:a...@ast-server.axvoice.com:5678;tag=5d297f22 Call-ID: MTE5N2M4ZDY1OWRjOGQwMjgyOWEzZjkzYjA3Y2RkYWY. CSeq: 1 SUBSCRIBE Expires: 300 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
[asterisk-users] How to extract channel-id of a user or peer
Hi,
my question is how can i get channel-id of a user or peer. I tried using
ChanIsAvail(username). this works correctly when user and asterisk are on
Local LAN. But my asterisk server is on public ip and users are behind nat,
and so this method says unknow host when used on public asterisk server.
I also tried built-in variable ${CHANNEL}, but this returns the channel-id
of the calling channel. but i want channel-id of called user.
can anyone help what can i do.
best regards,
Nasir Javaid
--
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[asterisk-users] Nat issue one way audio on IP dial
.x.x set_destination: Parsing sip:a...@116.18.35.235:28614 for address/port to send to set_destination: set destination to 116.18.35.235, port 28614 Transmitting (NAT) to 116.18.35.235:28614: ACK sip:a...@116.18.35.235:28614 SIP/2.0 Via: SIP/2.0/UDP 79.80.x.x:5678;branch=z9hG4bK07eb06b5;rport From: pepsi coke sip:12345678...@79.80.x.x:5678;tag=as42ec768c To: sip:a...@116.18.35.235:28614;tag=d54e632c Contact: sip:12345678...@79.80.x.x:5678 Call-ID: 0433af7878e3a8067a40f896382cc...@79.80.x.x CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- -- Call on SIP/116.18.35.235:28614-007f4660 left from hold -- SIP/116.18.35.235:28614-007f4660 answered SIP/pepsi-9fdfb9a0 ast-server*CLI --- SIP read from 116.18.35.235:28614 --- - --- (0 headers 1 lines) --- ast-server*CLI --- SIP read from 116.18.35.235:28614 --- - --- (0 headers 1 lines) --- Scheduling destruction of SIP dialog '0433af7878e3a8067a40f896382cc...@79.80.x.x' in 32000 ms (Method: INVITE) [Jul 21 11:11:03] DEBUG[23814]: chan_sip.c:5695 reqprep: Strict routing enforced for session 0433af7878e3a8067a40f896382cc...@79.80.x.x set_destination: Parsing sip:a...@116.18.35.235:28614 for address/port to send to set_destination: set destination to 116.18.35.235, port 28614 Reliably Transmitting (NAT) to 116.18.35.235:28614: BYE sip:a...@116.18.35.235:28614 SIP/2.0 Via: SIP/2.0/UDP 79.80.x.x:5678;branch=z9hG4bK36df65b5;rport From: pepsi coke sip:12345678...@79.80.x.x:5678;tag=as42ec768c To: sip:a...@116.18.35.235:28614;tag=d54e632c Call-ID: 0433af7878e3a8067a40f896382cc...@79.80.x.x CSeq: 103 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 Nasir Javaid -- _ -- 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] what is rinstance parameter in sip header
hello i was wondering what is the use of rinstance in SIP Headers. I noticed that this parameter is visible only when there is NAT invloved. I am experiencing one way audio when dialing a registered user by his IP:port. I this case rinstance parameter is missing. when i dial SIP/username audio is fine but when i dial SIP/x.x.x.x:port there is one way audion. Also please tell me what can go wrong by dialing by ip:port.?? Best regards, Nasir Javaid -- _ -- 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 URI Dial has one way audio
Hi, I am trying to dial a sip user via his IP:PORT Combination. i am using XYZ as target user which is registered. Asterisk server IP: 70.118.x.x calling user IP: 117.58.x.x called user IP:117.58.x.x:5062 First I dialed my registered user in normal way like this, Dial(SIP/XYZ,30,rtT) and during conversation audio was OK in both ways. Then I dialed the registered user via it's ip and port to which it was registered. like this, Dial(SIP/x...@117.58.x.x:5062,30,rtT) during conversation audio was one way just like before (calling party can hear called party but called party can not hear calling). after taking debug trace of both methods what I found was that a SIP HEADER parameter rinstance was missing in to and INVITE header fields when dialing via IP:PORT. I think this parameter is assigned automatically by asterisk. *NORMAL DIAL * INVITE sip:x...@:28614;rinstance=0266b8b94f488588 SIP/2.0 To: sip:x...@:28614;rinstance=0266b8b94f488588 Contact: sip:1334225...@xxx:5060 *IP DIAL* INVITE sip:x...@xxx:28614 SIP/2.0 To: sip:x...@:28614 Contact: sip:1334225...@xxx:5060 Is there something to be done with rinstance ?? 1) how can we assign this parameter when dialing via IP:PORT? 2) what else options do we have if we want to dial using IP:PORT mechanism. waiting for your kind response. Nasir Javaid. -- _ -- 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] One way audio when dialing multiple registrations
Hi again today when i was doing my research on this issue i found that even dialing a sip user by it's IP also raises this problem. here is what i did, First I dialed my registered user in normal way like this, Dial(SIP/XYZ,30,rtT) and during conversation audio was OK in both ways. Then I dialed the registered user via it's ip and port to which it was registered. like this, Dial(SIP/x...@:5062,30,rtT) during conversation audio was one way just like before (calling party can hear called party but called party can not hear calling). after taking debug trace of both methods what I found was that a SIP HEADER parameter rinstance was missing in to and INVITEt header fields when dialing via IP:PORT. I think this parameter is assigned automatically by asterisk. *NORMAL DIAL * INVITE sip:x...@:28614;rinstance=0266b8b94f488588 SIP/2.0 To: sip:x...@:28614;rinstance=0266b8b94f488588 Contact: sip:1334225...@xxx:5060 *IP DIAL* INVITE sip:x...@xxx:28614 SIP/2.0 To: sip:x...@:28614 Contact: sip:1334225...@xxx:5060 hope this research will ease a bit the quest to find a solution. now question is 1) how can we assign this parameter when dialing via IP:PORT? 2) what else options do we have if we want to dial using IP:PORT mechanism. waiting for your kind resopnse. Nasir Javaid. --- --- sorry for the typo mistake. the actual dial string that I used is like this Dial(SIP/x...@:5062-096afee8,30,rtT) Dial(SIP/x...@:64290-0966ab80,30,rtT) it is not Dial(SIP/192.168.0.20:5062-096afee8,30,rtT) Dial(SIP/192.168.0.12:64290-0966ab80,30,rtT) it was just a typing mistake that may have diverted all of you. hope this clears what i am trying to do. regards, Nasir Javaid --- I am sure you can't achieve what you are trying to achieve here. Simply use two different extensions instead of one. Considering how SIP communication works, I believe SIP doesn't allow multiple registrations like this. Maybe somebody can correct me here if I am wrong. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-07-19 12:28 PM, Nasir Javaid nasirjavaidna...@x wrote: thanks a lot zishan and philipp, probably that is the problem that is occurring. I am gonna take some wireshark or etherial trace to further investigate the problem. i don't wanna stuck into port forwarding issue as it will waste lot of time and also normal calling is working on my current port forwarding. what i am currently trying to grab the channel name along with it's unique id and dial it directly like simple Dial(SIP/xyz ) dialing for example Dial(SIP/192.168.0.20:5062-096afee8,30,rtT) Dial(SIP/192.168.0.12:64290-0966ab80,30,rtT) ^ | | | but problem is that asterisk assigns random unique-id for every call. and also it is available only when dialing... what are my options? your help will be highly appreciated. regards, Naisr Javaid - Based on the info you provided (though wireshark analysis will tell more about it), I am sure what is happening is that rtp coming back from the called doesn't know which ip to go to, because asterisk knows two ip addressses for the same extension due to the way you dialed it, i.e. in ringgroup fashion I have had this problem once and I never tried registering same extension from two different places after that. Try Phillip's suggestion, maybe it'll work for you. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-07-15 11:42 AM, Philipp von Klitzing klitz...@xx wrote: Hi! I am working on calling 2 registrations of same user on 2 different ip or ports. It works f... You need to make sure that these two phones use *different* RTP ports, and that this is handled correctly in your router/NAT device (by port forwarding or other methods). Philipp --- Hi Zeeshan, I saw many of your posts on forum. i also put my problem on forum but did not get any satisfying answer. I wish if you could help me out. below is my post
[asterisk-users] One way audio when dialing multiple registrations
sorry for the typo mistake. the actual dial string that I used is like this Dial(SIP/x...@192.168.0.20:5062-096afee8,30,rtT) Dial(SIP/x...@192.168.0.12:64290-0966ab80,30,rtT) it is not Dial(SIP/192.168.0.20:5062-096afee8,30,rtT) Dial(SIP/192.168.0.12:64290-0966ab80,30,rtT) it was just a typing mistake that may have diverted all of you. hope this clears what i am trying to do. regards, Nasir Javaid --- I am sure you can't achieve what you are trying to achieve here. Simply use two different extensions instead of one. Considering how SIP communication works, I believe SIP doesn't allow multiple registrations like this. Maybe somebody can correct me here if I am wrong. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-07-19 12:28 PM, Nasir Javaid nasirjavaidna...@x wrote: thanks a lot zishan and philipp, probably that is the problem that is occurring. I am gonna take some wireshark or etherial trace to further investigate the problem. i don't wanna stuck into port forwarding issue as it will waste lot of time and also normal calling is working on my current port forwarding. what i am currently trying to grab the channel name along with it's unique id and dial it directly like simple Dial(SIP/xyz ) dialing for example Dial(SIP/192.168.0.20:5062-096afee8,30,rtT) Dial(SIP/192.168.0.12:64290-0966ab80,30,rtT) ^ | | | but problem is that asterisk assigns random unique-id for every call. and also it is available only when dialing... what are my options? your help will be highly appreciated. regards, Naisr Javaid - Based on the info you provided (though wireshark analysis will tell more about it), I am sure what is happening is that rtp coming back from the called doesn't know which ip to go to, because asterisk knows two ip addressses for the same extension due to the way you dialed it, i.e. in ringgroup fashion I have had this problem once and I never tried registering same extension from two different places after that. Try Phillip's suggestion, maybe it'll work for you. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-07-15 11:42 AM, Philipp von Klitzing klitz...@xx wrote: Hi! I am working on calling 2 registrations of same user on 2 different ip or ports. It works f... You need to make sure that these two phones use *different* RTP ports, and that this is handled correctly in your router/NAT device (by port forwarding or other methods). Philipp --- Hi Zeeshan, I saw many of your posts on forum. i also put my problem on forum but did not get any satisfying answer. I wish if you could help me out. below is my post. == == Hi, I am working on calling 2 registrations of same user on 2 different ip or ports. It works fine and both phones ring simultaneously. the problem is that there is one way audio, calling party can hear me but i can't hear calling party. here is the scenario.. SIP/x...@xxx:5060 SIP/x...@:5678 i dial using following dial string Dial( SIP/x...@xxx:5060 SIP/x...@:5678,30,tTog) both destinations ring at the same time and one that is answered starts conversations. but audio is one sided as i mentioned above. But simply dialing single registration of XYZ like Dial(SIP/XYZ,30,tTog) works fine and audio is fine at both ends. have any idea what is going wrong?? any help will be highly appreciated regards, Nasir Javaid == thanks in advance ... -- _ -- 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] asterisk-users Digest, Vol 72, Issue 49
sorry for typo mistake in my last post. as from my orignal post two registration of the same user are as follows SIP/x...@119.68.0.90:5060 SIP/x...@202.16.34.10:5678 so dial command with unique-id i want to use will be Dial(SIP/x...@192.168.0.20:5062-096afee8,30,rtT) Dial(SIP/x...@192.168.0.12:64290-0966ab80,30,rtT) and not Dial(SIP/192.168.0.20:5062-096afee8,30,rtT) Dial(SIP/192.168.0.12:64290-0966ab80,30,rtT) now what you say is it still impossible? also can you explain what you mean by extension. username that i am using (XYZ) is registered at both ips and we dial 22129889035 which is associated with XYZ. like below user name: XYZ extension associated with XYZ: 22129889035 registered at:SIP/x...@192.168.0.20:5062and SIP/XYZ@ 192.168.0.12:64290 do you mean i use 2 different usernames or assign 2 different extensions to same user? as we already have 2-5 extension numbers associated with each username other than 22129889035. i attached sip debug trace but it was too heavy to be posted. if you say i can try posting it. is there any way for setting rtp port in dialplan. using functions like sipAddheader etc. so that i can set rtp ports with the channels involved in conversation at runtime. sincere regards, Nasir Javaid --- Message: 2 Date: Mon, 19 Jul 2010 13:41:32 -0400 From: Zeeshan Zakaria zisha...@gmail.com Subject: Re: [asterisk-users] One way audio when dialing multiple registrations To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: aanlktikou9eec7pfl2csxi83oyg0hnhzn34jecnqu...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 I am sure you can't achieve what you are trying to achieve here. Simply use two different extensions instead of one. Considering how SIP communication works, I believe SIP doesn't allow multiple registrations like this. Maybe somebody can correct me here if I am wrong. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-07-19 12:28 PM, Nasir Javaid nasirjavaidna...@gmail.com wrote: thanks a lot zishan and philipp, probably that is the problem that is occurring. I am gonna take some wireshark or etherial trace to further investigate the problem. i don't wanna stuck into port forwarding issue as it will waste lot of time and also normal calling is working on my current port forwarding. what i am currently trying to grab the channel name along with it's unique id and dial it directly like simple Dial(SIP/xyz ) dialing for example Dial(SIP/192.168.0.20:5062- 096afee8,30,rtT) Dial(SIP/192.168.0.12:64290-0966ab80,30,rtT) ^ | | | but problem is that asterisk assigns random unique-id for every call. and also it is available only when dialing... what are my options? your help will be highly appreciated. regards, Naisr Javaid - Based on the info you provided (though wireshark analysis will tell more about it), I am sure what is happening is that rtp coming back from the called doesn't know which ip to go to, because asterisk knows two ip addressses for the same extension due to the way you dialed it, i.e. in ringgroup fashion I have had this problem once and I never tried registering same extension from two different places after that. Try Phillip's suggestion, maybe it'll work for you. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-07-15 11:42 AM, Philipp von Klitzing klitz...@xx wrote: Hi! I am working on calling 2 registrations of same user on 2 different ip or ports. It works f... You need to make sure that these two phones use *different* RTP ports, and that this is handled correctly in your router/NAT device (by port forwarding or other methods). Philipp --- Hi Zeeshan, I saw many of your posts on forum. i also put my problem on forum but did not get any satisfying answer. I wish if you could help me out. below is my post. == == Hi, I am working on calling 2 registrations of same user on 2 different ip or ports. It works fine and both phones ring simultaneously. the problem is that there is one way audio, calling party can hear me but i can't hear calling party. here is the scenario.. SIP/x...@119.68.0.90:5060 SIP/x...@202.16.34.10:5678 i dial using following dial string Dial( SIP/x
[asterisk-users] One way audio when dialing multiple registrations
thanks a lot zishan and philipp, probably that is the problem that is occurring. I am gonna take some wireshark or etherial trace to further investigate the problem. i don't wanna stuck into port forwarding issue as it will waste lot of time and also normal calling is working on my current port forwarding. what i am currently trying to grab the channel name along with it's unique id and dial it directly like simple Dial(SIP/xyz ) dialing for example Dial(SIP/192.168.0.20:5062-096afee8,30,rtT) Dial(SIP/192.168.0.12:64290-0966ab80,30,rtT) ^ | | | but problem is that asterisk assigns random unique-id for every call. and also it is available only when dialing... what are my options? your help will be highly appreciated. regards, Naisr Javaid - Based on the info you provided (though wireshark analysis will tell more about it), I am sure what is happening is that rtp coming back from the called doesn't know which ip to go to, because asterisk knows two ip addressses for the same extension due to the way you dialed it, i.e. in ringgroup fashion I have had this problem once and I never tried registering same extension from two different places after that. Try Phillip's suggestion, maybe it'll work for you. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-07-15 11:42 AM, Philipp von Klitzing klitz...@xx wrote: Hi! I am working on calling 2 registrations of same user on 2 different ip or ports. It works f... You need to make sure that these two phones use *different* RTP ports, and that this is handled correctly in your router/NAT device (by port forwarding or other methods). Philipp --- Hi Zeeshan, I saw many of your posts on forum. i also put my problem on forum but did not get any satisfying answer. I wish if you could help me out. below is my post. == == Hi, I am working on calling 2 registrations of same user on 2 different ip or ports. It works fine and both phones ring simultaneously. the problem is that there is one way audio, calling party can hear me but i can't hear calling party. here is the scenario.. SIP/x...@119.68.0.90:5060 SIP/x...@202.16.34.10:5678 i dial using following dial string Dial( SIP/x...@119.68.0.90:5060 SIP/x...@202.16.34.10:5678,30,tTog) both destinations ring at the same time and one that is answered starts conversations. but audio is one sided as i mentioned above. But simply dialing single registration of XYZ like Dial(SIP/XYZ,30,tTog) works fine and audio is fine at both ends. have any idea what is going wrong?? any help will be highly appreciated regards, Nasir Javaid == thanks in advance ... -- _ -- 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] asterisk-users Digest, Vol 72, Issue 39
yes, actually this scenario is on remote servers. like SIP/x...@119.18.230.20:5060 SIP/x...@202.68.0.90:5678 audio is ok when dialing without using ip port as below SIP/XYZ but when i dial using below dialstring SIP/x...@202.68.0.90:5678 or SIP/x...@119.18.230.20:5060 then the problem arises hope you got the idea.. Nasir -- Message: 26 Date: Thu, 15 Jul 2010 17:09:06 +0200 From: Jonas Kellens jonas.kell...@telenet.be Subject: Re: [asterisk-users] One way audio when dialing multiple registrations To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: 4c3f2492.4040...@telenet.be Content-Type: text/plain; charset=iso-8859-1 One-way audio is mostly firewall problem. Are you behind firewall ? You can check the audio-ports that are being used in the SDP-message by doing a /sip debug/. Maybe you do not have enough UDP-ports open for the audio ? Jonas. On 07/15/2010 04:38 PM, Nasir Javaid wrote: Hi, I am working on calling 2 registrations of same user on 2 different ip or ports. It works fine and both phones ring simultaneously. the problem is that there is one way audio, calling party can hear me but i can't hear calling party. here is the scenario.. SIP/x...@192.168.0.20:5060 http://x...@192.168.0.20:5060 SIP/x...@192.168.0.10:5678 http://x...@192.168.0.10:5678 i dial using following dial string Dial(SIP/x...@192.168.0.20:5060SIP/x...@192.168.0.10:5678 http://x...@192.168.0.10:5678,30,tTog) both destinations ring at the same time and one that is answered starts conversations. but audio is one sided as i mentioned above. But simply dialing single registration of XYZ like Dial(SIP/XYZ,30,tTog) works fine and audio is fine at both ends. have any idea what is going wrong?? any help will be highly appreciated regards, Nasir Javaid -- next part -- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100715/b51fed83/attachment-0001.htm -- -- _ -- 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] One way audio when dialing multiple registrations
Hi, I am working on calling 2 registrations of same user on 2 different ip or ports. It works fine and both phones ring simultaneously. the problem is that there is one way audio, calling party can hear me but i can't hear calling party. here is the scenario.. SIP/x...@192.168.0.20:5060 SIP/x...@192.168.0.10:5678 i dial using following dial string Dial(SIP/x...@192.168.0.20:5060SIP/x...@192.168.0.10:5678,30,tTog) both destinations ring at the same time and one that is answered starts conversations. but audio is one sided as i mentioned above. But simply dialing single registration of XYZ likeDial(SIP/XYZ,30,tTog) works fine and audio is fine at both ends. have any idea what is going wrong?? any help will be highly appreciated regards, Nasir Javaid -- _ -- 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] asterisk-users Digest, Vol 70, Issue 54
Hi, I am having very strange situation. I have my sip peer located over the internet and I am able to connect and dial to it. the problem is that, if yesterday i was connected with it via my asterisk client and dialing to it normal way, today when i run asterisk on my client, my sip peer becomes unreachable. there is no change in settings or anything else. it may become reachable after one hour or one day. but all this is random. similarly it could be reachable from one client and unreachable from other client on the LAN. can anyone help me out what is going wrong. I think this could be network issue but don't know how to prove it thanks Nasir Javaid -- _ -- 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] asterisk-users Digest, Vol 70, Issue 30
sorry, you r right i just checked it with registration so there were astdb
entries for SIP registration.
anyhow after clearing settings frm astdb i tried the same scenario you
advised but no luck.
I think i told that i am not using server as peer but want to use a user
[abc] as peer so that when ever i use
dial(SIP/${ext...@abc) or dial(SIP/abc/${EXTEN})
the call will be out from server using [abc]'s account.
i hope you understand what i mean.
also i will like to know is there any way that i can include registration
information in my dial string so that i have no need to write
register = abc:mysec...@nasir.server.com:8060
regards,
Nasir Javaid
Look, you do again with registration.
remove any registration information.
Look this config, I think it can help you
Server1:
sip.conf
[interboxserver2]
type=friend
host=192.168.250.112
context=callfromserver2
disallow=all
allow=ulaw
allow=alaw
allow=g729
extensions.conf
[calltoserver2]
exten = _X.,1,Noop(Call to server2)
exten = _X.,2,Dial(SIP/
interboxserver2/${EXTEN})
exten = _X.,3,Hangup
[callfromserver2]
exten = _X.,1,Noop(Call from server2)
exten = _X.,2,Dial(SIP/${EXTEN})
exten = _X.,3,Hangup
Server2:
sip.conf
[interboxserver1]
type=friend
host=192.168.250.111
context=callfromserver1
disallow=all
allow=ulaw
allow=alaw
allow=g729
extensions.conf
[calltoserver1]
exten = _X.,1,Noop(Call to server1)
exten = _X.,2,Dial(SIP/interboxserver1/${EXTEN})
exten = _X.,3,Hangup
[callfromserver1]
exten = _X.,1,Noop(Call from server1)
exten = _X.,2,Dial(SIP/${EXTEN})
exten = _X.,3,Hangup
Try so, I think it must work.
And also, look and delete any another records in both servers in
sip.conf about this servers settings.
Vardan
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Re: [asterisk-users] asterisk-users Digest, Vol 70, Issue 25
Hi Vardan
I did same as you told and deleted the SIP information in Astdb and
restarted asterisk. but the result was same.
as you said there might be mistake in sip.conf so i am pasting both servers
configuration here..
1- nasir.server.com
[abc]
username=abc
type=friend
secret=mysecret
nat=yes
mailbox=12234568
incominglimit=2
outgoinglimit=2
host=dynamic
dtmfmode=rfc2833
context=payasyougo
canreinvite=yes
callerid=Nasir Qazi 12234
accountcode=6:0:abc
amaflags=default
disallow=all
allow=ulaw
allow=alaw
allow=g729
allow=gsm
2- 192.168.0.254 (client system)
[abc]
type=peer
username=abc
secret=mysecret
host=nasir.server.com
context=default
dtmfmode=rfc2833
canreinvite=yes
insecure=very
disallow=all
allow=ulaw
allow=alaw
allow=g729
allow=gsm
nat=yes
;qualify=yes
[caller]
type=friend
secret=123456
host=dynamic
callerid=caller 1212988
context=out
nat=yes
dtmfmode=rfc2833
canreinvite=yes
insecure=no
disallow=all
allow=ulaw
allow=alaw
allow=g729
allow=gsm
t38_udptl=yes
qualify=yes
I have registered [caller] on xlite at client system and dialing following
context in local system that will dial [abc]
[out]
exten= _X.,1,Dial(SIP/${ext...@abc,30,1)
exten= _X.,n,Hangup
as you can see above *highlighted that context of abc is
payasyougo.*problem is that i want the call to land in that context on
nasir.server.com, which works if i use register string. but without register
string call goes to default context on nasir.server.com
regards,
Nasir Javaid
Message: 19
Date: Tue, 11 May 2010 20:54:30 +0500
From: Vardan hvarda...@gmail.com
Subject: Re: [asterisk-users] asterisk-users Digest, Vol 70, Issue 24
To: asterisk-users@lists.digium.com
Message-ID: hsbujk$qk...@dough.gmane.org
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Hello Nasir
I have some please.
Do so, it help.
Find all records about interexchange beetwen this two server and delete
all records in sip.conf for this both server (first make backup
sip.conf, or any another conf file that you use).
restart asterisk.
look in astdb about this old records, if any found, delete him
Next, create new record in sip.conf on both servers, without
registration string, reload sip.conf.
give him right context from extensions.conf.
Can you do this?
I think is some mistake about configuration in sip.conf, you have I
think two same records (peer or friend).
Vardan
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Re: [asterisk-users] asterisk-users Digest, Vol 70, Issue 25
)
[May 12 19:21:14] Peer audio RTP is at port 192.168.0.254:11026
[May 12 19:21:14] Looking for 17185594743 in payasyougo (domain
nasir.server.com)
[May 12 19:21:14] WARNING[3785]: chan_sip.c:3930 sip_new: setting callerid
number to 12129339037
[May 12 19:21:14] list_route: hop:
sip:12129887...@192.168.0.254sip%3a12129887...@192.168.0.254
[May 12 19:21:14]
--- Transmitting (NAT) to 192.168.0.254:5060 ---
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.254:5060
;branch=z9hG4bK05611806;received=192.168.0.254;rport=5060
From: caller sip:12129887...@192.168.0.254sip%3a12129887...@192.168.0.254
;tag=as76623e31
To: sip:17185594...@nasir.server.com sip%3a17185594...@nasir.server.com
Call-ID: 245c407103141a6841c0ac106bd5a...@192.168.0.254
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:17185594...@nasir.server.comsip%3a17185594...@nasir.server.com
Content-Length: 0
On Wed, May 12, 2010 at 7:14 PM, Nasir Javaid nasirjavaidna...@gmail.comwrote:
Hi Vardan
I did same as you told and deleted the SIP information in Astdb and
restarted asterisk. but the result was same.
as you said there might be mistake in sip.conf so i am pasting both servers
configuration here..
1- nasir.server.com
[abc]
username=abc
type=friend
secret=mysecret
nat=yes
mailbox=12234568
incominglimit=2
outgoinglimit=2
host=dynamic
dtmfmode=rfc2833
context=payasyougo
canreinvite=yes
callerid=Nasir Qazi 12234
accountcode=6:0:abc
amaflags=default
disallow=all
allow=ulaw
allow=alaw
allow=g729
allow=gsm
2- 192.168.0.254 (client system)
[abc]
type=peer
username=abc
secret=mysecret
host=nasir.server.com
context=default
dtmfmode=rfc2833
canreinvite=yes
insecure=very
disallow=all
allow=ulaw
allow=alaw
allow=g729
allow=gsm
nat=yes
;qualify=yes
[caller]
type=friend
secret=123456
host=dynamic
callerid=caller 1212988
context=out
nat=yes
dtmfmode=rfc2833
canreinvite=yes
insecure=no
disallow=all
allow=ulaw
allow=alaw
allow=g729
allow=gsm
t38_udptl=yes
qualify=yes
I have registered [caller] on xlite at client system and dialing following
context in local system that will dial [abc]
[out]
exten= _X.,1,Dial(SIP/${ext...@abc,30,1)
exten= _X.,n,Hangup
as you can see above *highlighted that context of abc is payasyougo.*problem
is that i want the call to land in that context on
nasir.server.com, which works if i use register string. but without
register string call goes to default context on nasir.server.com
regards,
Nasir Javaid
Message: 19
Date: Tue, 11 May 2010 20:54:30 +0500
From: Vardan hvarda...@gmail.com
Subject: Re: [asterisk-users] asterisk-users Digest, Vol 70, Issue 24
To: asterisk-users@lists.digium.com
Message-ID: hsbujk$qk...@dough.gmane.org
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Hello Nasir
I have some please.
Do so, it help.
Find all records about interexchange beetwen this two server and delete
all records in sip.conf for this both server (first make backup
sip.conf, or any another conf file that you use).
restart asterisk.
look in astdb about this old records, if any found, delete him
Next, create new record in sip.conf on both servers, without
registration string, reload sip.conf.
give him right context from extensions.conf.
Can you do this?
I think is some mistake about configuration in sip.conf, you have I
think two same records (peer or friend).
Vardan
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Re: [asterisk-users] asterisk-users Digest, Vol 70, Issue 25
Hi again, below is debug trace of * cli when i remove register string from sip.conf *CLI [May 12 19:33:06] --- SIP read from 192.168.0.254:5060 --- INVITE sip:17185594...@nasir.server.com sip%3a17185594...@nasir.server.comSIP/2.0 Via: SIP/2.0/UDP 192.168.0.254:5060;branch=z9hG4bK56e3b44a;rport Max-Forwards: 70 From: caller sip:12129887...@192.168.0.254sip%3a12129887...@192.168.0.254 ;tag=as5b6db7a2 To: sip:17185594...@nasir.server.com sip%3a17185594...@nasir.server.com Contact: sip:12129887...@192.168.0.254 sip%3a12129887...@192.168.0.254 Call-ID: 23c4c49b329104d31ad6822c02cb8...@192.168.0.254 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.2.0 Date: Wed, 12 May 2010 14:32:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 284 v=0 o=root 814806874 814806874 IN IP4 192.168.0.254 s=Asterisk PBX 1.6.2.0 c=IN IP4 192.168.0.254 t=0 0 m=audio 17632 RTP/AVP 0 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv - [May 12 19:33:06] --- (14 headers 13 lines) --- [May 12 19:33:06] Sending to 192.168.0.254 : 5060 (NAT) [May 12 19:33:06] Using INVITE request as basis request - 23c4c49b329104d31ad6822c02cb8...@192.168.0.254 [May 12 19:33:06] Found no matching peer or user for '192.168.0.254:5060' [May 12 19:33:06] Found RTP audio format 0 [May 12 19:33:06] Found RTP audio format 3 [May 12 19:33:06] Found RTP audio format 101 [May 12 19:33:06] Peer audio RTP is at port 192.168.0.254:17632 [May 12 19:33:06] Found description format PCMU for ID 0 [May 12 19:33:06] Found description format GSM for ID 3 [May 12 19:33:06] Found description format telephone-event for ID 101 [May 12 19:33:06] Capabilities: us - 0x10e (gsm|ulaw|alaw|g729), peer - audio=0x6 (gsm|ulaw)/video=0x0 (nothing), combined - 0x6 (gsm|ulaw) [May 12 19:33:06] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [May 12 19:33:06] Peer audio RTP is at port 192.168.0.254:17632 [May 12 19:33:06] Looking for 17185594743 in default (domain nasir.server.com) [May 12 19:33:06] WARNING[4113]: chan_sip.c:3930 sip_new: setting callerid number to 1212988 [May 12 19:33:06] list_route: hop: sip:12129887...@192.168.0.254sip%3a12129887...@192.168.0.254 [May 12 19:33:06] --- Transmitting (NAT) to 192.168.0.254:5060 --- SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.254:5060 ;branch=z9hG4bK56e3b44a;received=192.168.0.254;rport=5060 From: caller sip:12129887...@192.168.0.254sip%3a12129887...@192.168.0.254 ;tag=as5b6db7a2 To: sip:17185594...@nasir.server.com sip%3a17185594...@nasir.server.com Call-ID: 23c4c49b329104d31ad6822c02cb8...@192.168.0.254 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: sip:17185594...@nasir.server.comsip%3a17185594...@nasir.server.com Content-Length: 0 On Wed, May 12, 2010 at 7:26 PM, Nasir Javaid nasirjavaidna...@gmail.comwrote: here i am attaching debug trace of sip in case of sccessfull call when using register string... *CLI [May 12 19:21:14] --- SIP read from 192.168.0.254:5060 --- INVITE sip:17185594...@nasir.server.comsip%3a17185594...@nasir.server.comSIP/2.0 Via: SIP/2.0/UDP 192.168.0.254:5060;branch=z9hG4bK3c63f272;rport Max-Forwards: 70 From: caller sip:12129887...@192.168.0.254sip%3a12129887...@192.168.0.254 ;tag=as76623e31 To: sip:17185594...@nasir.server.com sip%3a17185594...@nasir.server.com Contact: sip:12129887...@192.168.0.254 sip%3a12129887...@192.168.0.254 Call-ID: 245c407103141a6841c0ac106bd5a...@192.168.0.254 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.2.0 Date: Wed, 12 May 2010 14:20:25 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 284 v=0 o=root 618893758 618893758 IN IP4 192.168.0.254 s=Asterisk PBX 1.6.2.0 c=IN IP4 192.168.0.254 t=0 0 m=audio 11026 RTP/AVP 0 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv - [May 12 19:21:14] --- (14 headers 13 lines) --- [May 12 19:21:14] Sending to 192.168.0.254 : 5060 (NAT) [May 12 19:21:14] Using INVITE request as basis request - 245c407103141a6841c0ac106bd5a...@192.168.0.254 [May 12 19:21:14] Found peer 'abc' [May 12 19:21:14] --- Reliably Transmitting (NAT) to 192.168.0.254:5060 --- SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.0.254:5060 ;branch=z9hG4bK3c63f272;received=192.168.0.254;rport=5060 From: caller sip:12129887...@192.168.0.254sip%3a12129887...@192.168.0.254 ;tag=as76623e31 To: sip:17185594...@nasir.server.com sip%3a17185594...@nasir.server.com ;tag=as0a721b3a Call-ID: 245c407103141a6841c0ac106bd5a...@192.168.0.254 CSeq
Re: [asterisk-users] asterisk-users Digest, Vol 70, Issue 23
Thanks Vardan, I will like to know if this scenario can work when peer is not having fixed ip and we use host = nasir.server.com ? also I have set insecure=invite,port what if i use insecure=no thanks again. Message: 24 Date: Tue, 11 May 2010 10:52:14 +0500 From: Vardan hvarda...@gmail.com Subject: Re: [asterisk-users] Dialing a SIP Peer without using register strin To: asterisk-users@lists.digium.com Message-ID: hsarab$ok...@dough.gmane.org Content-Type: text/plain; charset=ISO-8859-1; format=flowed Remove username and secret and use IP authentication on both side [server1_abc] type=peer host=192.168.0.20 context=default dtmfmode=rfc2833 canreinvite=yes - canreinvite with nat=yes is not working insecure=invite,port disallow=all allow=ulaw allow=alaw allow=g729 allow=gsm nat=yes qualify=yes [server2_abc] type=peer host=192.168.0.21 context=default dtmfmode=rfc2833 canreinvite=yes insecure=invite,port disallow=all allow=ulaw allow=alaw allow=g729 allow=gsm nat=yes qualify=yes Nasir Javaid wrote: Hi, I am new to this list and this is first time i m posting here. please help me out currently I am working on dialing a sip peer on an asterisk server from 2nd asterisk server. scenario is like this. on my system i am using this peer in sip.conf. [abc] type=peer username=abc secret=mysecret host=192.168.0.20 context=default dtmfmode=rfc2833 ;restrictcid=no canreinvite=yes insecure=invite,port disallow=all allow=ulaw allow=alaw allow=g729 allow=gsm nat=yes qualify=yes and using following register string register = abc:mysec...@192.168.0.20 abc%3amysec...@192.168.0.20mailto: abc%3amysec...@192.168.0.20 abc%253amysec...@192.168.0.20 now problem is that when i use register string everything goes ok. but when i remove register string call doesn't go as expected. I would like to know if there is any feature that i can use to call sip peer and authenticate is in dial command or some feature in sip.conf i dont wanna use register string. please help. regards, Nasir Javaid -- _ -- 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] asterisk-users Digest, Vol 70, Issue 24
Yes this scenario works on my 2 systems which are at LAN. I made one system
as server (192.168.0.20) and registered from other system... it is fine but
now there is a different scene.
actually there is a registered user named abc at system1 (192.168.0.20)
having context [payasyougo] which is used to do outbound calls. we want to
use this user's context and account so that when we register this account on
our other system (192.168.0.254) call should be dropped into the context of
abc which is [payasyougo].
now this works fine on above mentioned systems, and calling from system
(192.168.0.254) like this
SIP/${ext...@abc
sends call to the abc's context [payasyougo] and from there system1
(192.168.0.20) takes charge of dialing out the number in ${EXTEN}.
but when i change system 1 (192.168.0.20) to my real server (e.g.
nasir.server.com) which has abc as user configured same as on system1
(192.168.0.20), call goes to [default] instead of going to [payasyougo]
context and is treated as incoming call...
when we use register string calls works ok on real server too. I also tried
SIP/abc:mysecret/${EXTEN} and
SIP/${ext...@abc:mysecret
but nothing seems to work.
there is another problem that sometime my real server (nasir.server.com)
becomes unreachable and this error is returned
NOTICE[3898]: chan_sip.c:11489 sip_reg_timeout:-- Registration for '
a...@nasir.server.com' timed out, trying again (Attempt #38)
It may be a simple problem but is driving my crazy... please help me out
thanks in advance
Nasir Javaid
Message: 6
Date: Tue, 11 May 2010 13:57:23 +0500
From: Nasir Javaid nasirjavaidna...@gmail.com
Subject: Re: [asterisk-users] asterisk-users Digest, Vol 70, Issue 23
To: asterisk-users@lists.digium.com
Message-ID:
aanlktim3qspcy0sy3handxttg8hpbxjemj7hwzskn...@mail.gmail.com
Content-Type: text/plain; charset=iso-8859-1
Thanks Vardan,
I will like to know if this scenario can work when peer is not having fixed
ip and we use
host = nasir.server.com
?
also I have set insecure=invite,port
what if i use
insecure=no
thanks again.
Message: 24
Date: Tue, 11 May 2010 10:52:14 +0500
From: Vardan hvarda...@gmail.com
Subject: Re: [asterisk-users] Dialing a SIP Peer without using
register strin
To: asterisk-users@lists.digium.com
Message-ID: hsarab$ok...@dough.gmane.org
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Remove username and secret and use IP authentication on both side
[server1_abc]
type=peer
host=192.168.0.20
context=default
dtmfmode=rfc2833
canreinvite=yes - canreinvite with nat=yes is not working
insecure=invite,port
disallow=all
allow=ulaw
allow=alaw
allow=g729
allow=gsm
nat=yes
qualify=yes
[server2_abc]
type=peer
host=192.168.0.21
context=default
dtmfmode=rfc2833
canreinvite=yes
insecure=invite,port
disallow=all
allow=ulaw
allow=alaw
allow=g729
allow=gsm
nat=yes
qualify=yes
Nasir Javaid wrote:
Hi,
I am new to this list and this is first time i m posting here. please
help me out
currently I am working on dialing a sip peer on an asterisk server from
2nd asterisk server. scenario is like this.
on my system i am using this peer in sip.conf.
[abc]
type=peer
username=abc
secret=mysecret
host=192.168.0.20
context=default
dtmfmode=rfc2833
;restrictcid=no
canreinvite=yes
insecure=invite,port
disallow=all
allow=ulaw
allow=alaw
allow=g729
allow=gsm
nat=yes
qualify=yes
and using following register string
register = abc:mysec...@192.168.0.20 abc%3amysec...@192.168.0.20
abc%3amysec...@192.168.0.20 abc%253amysec...@192.168.0.20mailto:
abc%3amysec...@192.168.0.20 abc%253amysec...@192.168.0.20
abc%253amysec...@192.168.0.20 abc%25253amysec...@192.168.0.20
now problem is that when i use register string everything goes ok. but
when i remove register string call doesn't go as expected.
I would like to know if there is any feature that i can use to call sip
peer and authenticate is in dial command or some feature in sip.conf
i dont wanna use register string. please help.
regards,
Nasir Javaid
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[asterisk-users] Dialing a SIP Peer without using register strin
Hi, I am new to this list and this is first time i m posting here. please help me out currently I am working on dialing a sip peer on an asterisk server from 2nd asterisk server. scenario is like this. on my system i am using this peer in sip.conf. [abc] type=peer username=abc secret=mysecret host=192.168.0.20 context=default dtmfmode=rfc2833 ;restrictcid=no canreinvite=yes insecure=invite,port disallow=all allow=ulaw allow=alaw allow=g729 allow=gsm nat=yes qualify=yes and using following register string register = abc:mysec...@192.168.0.20 abc%3amysec...@192.168.0.20 now problem is that when i use register string everything goes ok. but when i remove register string call doesn't go as expected. I would like to know if there is any feature that i can use to call sip peer and authenticate is in dial command or some feature in sip.conf i dont wanna use register string. please help. regards, Nasir Javaid -- _ -- 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
