[asterisk-users] Function_CHANNEL how to get source ip address in dial plan?
Hi, I using PJSIP as sip driver, I wound like to get source IP on inbound calls
from my peers,
tried use Function_CHANNEL like
${CHANNEL(pjsip,type,remote_addr)}
but it returns only empty string, what I doing wrong?
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Re: [asterisk-users] Issues with call dropping
May someone help with the sourcecode, trying find where can I manually send response on Received INFO request in PJSIP ASTERISK-24986 issues opened already more the 2 month and calls from customers still drops. very annoying :( maybe some one could help me figure out where Received INFO request dies in source so I could patch it to response 200 OK ? > On 20 Apr 2015, at 15:08, Nick Awesome wrote: > > Hi guys, have really annoying problem with trunks when I calling over voip > provider.. > > > after awhile provider sends INFO packages but for some reason Asterisk > doesn’t answer on it. > after 8 packagers provider just drops the call, here is the package > > <--- Received SIP request (555 bytes) from UDP:192.168.53.1:5060 ---> > INFO sip:c2e74184-4d23-43a3-8fd9-26ba2c5ef6c9@192.168.53.9:5060 SIP/2.0 > Max-Forwards: 69 > To: ;tag=b3769af4-118b-4467-8c95-042247ff1776 > From: ;tag=3638518512-132845 > Call-ID: cfe34652-14c2-4072-9dea-f0b0c30cb15e > CSeq: 2 INFO > Allow: CANCEL, ACK, INVITE, BYE, OPTIONS, REGISTER, NOTIFY, INFO, REFER, > SUBSCRIBE, PRACK, UPDATE, MESSAGE, PUBLISH > Via: SIP/2.0/UDP > 192.168.53.1:5060;branch=z9hG4bK8da58cede20b91eb54dec15ad27f866c > Contact: > Content-Length: 0 > > 192.168.53.1 - operator IP > 192.168.53.9 - asterisk IP > > > Any idea how to fix this? > > > have 2 Ethernet interfaces: > 192.168.1.4 - local network > 192.168.53.9 - VOIP Provider network > > Im using PJSIP, here is config: > > [udp] > type=transport > protocol=udp > bind=192.168.1.4 > local_net=10.0.0.0/24 > local_net=10.0.1.0/24 > local_net=192.168.1.0/24 > > external_media_address=195.239.8.122 > external_signaling_address=195.239.8.122 > > [udp_B] > type=transport > protocol=udp > bind=192.168.53.9 > > [1] > type=endpoint > aors=1 > context=dialmap > disallow=all > allow=alaw,ulaw > transport=udp_B > > [1] > type=aor > contact=sip:192.168.53.1:5060 > max_contacts=4 > -- _ -- 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] ARI echo test
recreate Echo, if that is possible. trying to recode all dialplan to stasis application > On 22 May 2015, at 19:29, Scott Griepentrog wrote: > > Nick- > > Are you wanting to recreate the dialplan Echo() application in stasis? > > Why not just send the call to Echo() instead of Stasis()? > > On Fri, May 22, 2015 at 11:25 AM, Matthew Jordan <mailto:mjor...@digium.com>> wrote: > On Fri, May 22, 2015 at 4:41 AM, Nick Awesome <mailto:jl...@me.com>> wrote: > > Can anyone tell me how can I create echo test using ARI stasis application? > > > > I'm not sure an 'echo' test really makes much sense with ARI, but we > do have some nice documentation on getting started with ARI on the > wiki. The basic tutorial example should give you an ARI event over a > WebSocket connection. > > https://wiki.asterisk.org/wiki/display/AST/Getting+Started+with+ARI > <https://wiki.asterisk.org/wiki/display/AST/Getting+Started+with+ARI> > > -- > Matthew Jordan > Digium, Inc. | Director of Technology > 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA > Check us out at: http://digium.com <http://digium.com/> & http://asterisk.org > <http://asterisk.org/> > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com > <http://www.api-digital.com/> -- > New to Asterisk? Join us for a live introductory webinar every Thurs: >http://www.asterisk.org/hello <http://www.asterisk.org/hello> > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > <http://lists.digium.com/mailman/listinfo/asterisk-users> > > > > -- > > Scott Griepentrog > Digium, Inc · Software Developer > 445 Jan Davis Drive NW · Huntsville, AL 35806 · US > direct/fax: +1 256 428 6239 · mobile: +1 256 580 6090 > Check us out at: http://digium.com <http://digium.com/> · http://asterisk.org > <http://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 smime.p7s Description: S/MIME cryptographic 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
[asterisk-users] ARI echo test
Can anyone tell me how can I create echo test using ARI stasis application? -- _ -- 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] getting lots of warnings
what may cause this, and how can I fix it ? WARNING[23010]: pjsip:0 : tsx0x7f24f41b2 ..Failed to send Request msg NOTIFY/cseq=15293 (tdta0x7f2480001a70)! err=171064 (Unsuitable transport selected (PJSIP_ETPNOTSUITABLE))-- _ -- 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] Issues with call dropping
Hi guys, have really annoying problem with trunks when I calling over voip provider.. after awhile provider sends INFO packages but for some reason Asterisk doesn’t answer on it. after 8 packagers provider just drops the call, here is the package <--- Received SIP request (555 bytes) from UDP:192.168.53.1:5060 ---> INFO sip:c2e74184-4d23-43a3-8fd9-26ba2c5ef6c9@192.168.53.9:5060 SIP/2.0 Max-Forwards: 69 To: ;tag=b3769af4-118b-4467-8c95-042247ff1776 From: ;tag=3638518512-132845 Call-ID: cfe34652-14c2-4072-9dea-f0b0c30cb15e CSeq: 2 INFO Allow: CANCEL, ACK, INVITE, BYE, OPTIONS, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE, MESSAGE, PUBLISH Via: SIP/2.0/UDP 192.168.53.1:5060;branch=z9hG4bK8da58cede20b91eb54dec15ad27f866c Contact: Content-Length: 0 192.168.53.1 - operator IP 192.168.53.9 - asterisk IP Any idea how to fix this? have 2 Ethernet interfaces: 192.168.1.4 - local network 192.168.53.9 - VOIP Provider network Im using PJSIP, here is config: [udp] type=transport protocol=udp bind=192.168.1.4 local_net=10.0.0.0/24 local_net=10.0.1.0/24 local_net=192.168.1.0/24 external_media_address=195.239.8.122 external_signaling_address=195.239.8.122 [udp_B] type=transport protocol=udp bind=192.168.53.9 [1] type=endpoint aors=1 context=dialmap disallow=all allow=alaw,ulaw transport=udp_B [1] type=aor contact=sip:192.168.53.1:5060 max_contacts=4 -- _ -- 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 switching bridge to native_rtp even with direct_media=no
NAT endpoint calling local endpount - switching to native_rtp then no audio,
both of them have direct_media=no, Verbose log:
-- Executing [99@dialmap:1] AGI("PJSIP/304-0022", "/pbx/agi.php") in
new stack
-- Launched AGI Script /pbx/agi.php
-- AGI Script Executing Application: (Dial) Options:
(PJSIP/99/sip:99@192.168.1.73:5060,20)
-- Called PJSIP/99/sip:99@192.168.1.73:5060
-- PJSIP/99-0023 is ringing
-- PJSIP/99-0023 answered PJSIP/304-0022
-- Channel PJSIP/304-0022 joined 'simple_bridge' basic-bridge
-- Channel PJSIP/99-0023 joined 'simple_bridge' basic-bridge
> Bridge da8840bc-9b71-4ca6-b1d8-9565bf8e5e28: switching from
simple_bridge technology to native_rtp
> Locally RTP bridged 'PJSIP/99-0023' and 'PJSIP/304-0022' in
stack
> Locally RTP bridged 'PJSIP/99-0023' and 'PJSIP/304-0022' in
stack
> 0x7f4b50145420 -- Probation passed - setting RTP source address to
194.204.157.200:8972
> 0x7f4b5014f140 -- Probation passed - setting RTP source address to
192.168.1.73:5004
-- Channel PJSIP/304-0022 left 'native_rtp' basic-bridge
-- Channel PJSIP/99-0023 left 'native_rtp' basic-bridge
-- AGI Script /pbx/agi.php completed, returning 4
> On 18 Mar 2015, at 18:26, Matthew Jordan wrote:
>
> On Wed, Mar 18, 2015 at 9:53 AM, Nick Awesome wrote:
>> Well, it breaks audio for all NAT endpoints, how can I fix this?
>>
>
> Local (packet to packet) bridging should not do that. Remote (direct
> media) can do that.
>
> Can you confirm - by looking at a verbose level 4 log - how Asterisk
> is bridging the two channels?
>
> --
> Matthew Jordan
> Digium, Inc. | Director of Technology
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> Check us out at: http://digium.com & http://asterisk.org
>
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> http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [asterisk-users] Asterisk switching bridge to native_rtp even with direct_media=no
Well, it breaks audio for all NAT endpoints, how can I fix this? > On 18 Mar 2015, at 15:48, Matthew Jordan wrote: > > On Wed, Mar 18, 2015 at 7:41 AM, Nick Awesome <mailto:jl...@me.com>> wrote: >> Hey guys, >> >> have issues with reinvite, no matter what endpoint is calling asterisk >> always tries switch simple_bridge to native_rtp >> >> Bridge 0422bfa0-9d22-4bba-9108-a3f14d7d1cab: switching from simple_bridge >> technology to native_rtp >> >> in endpoints table “direct_media” sets to “no” on all endpoints but it >> doesn’t help. >> >> if native_rtp not work for some reason I have oneway audio. how can I fix >> this? if I add mix_monitor it works, but it’s not a right way to fix this >> issues. >> > > A native_rtp bridge is used for more than direct media. It is also > used for local native bridging, that is, when you have two RTP capable > channels in a bridge and Asterisk does not require the media to flow > through its core. The bridge then just performs a packet to packet > swap between the two RTP capable channels. > > Note that on verbosity 4, Asterisk will tell you if the bridge is > locally or remotely bridging the two channels. > > -- > Matthew Jordan > Digium, Inc. | Director of Technology > 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA > Check us out at: http://digium.com <http://digium.com/> & http://asterisk.org > <http://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
[asterisk-users] Asterisk switching bridge to native_rtp even with direct_media=no
Hey guys, have issues with reinvite, no matter what endpoint is calling asterisk always tries switch simple_bridge to native_rtp Bridge 0422bfa0-9d22-4bba-9108-a3f14d7d1cab: switching from simple_bridge technology to native_rtp in endpoints table “direct_media” sets to “no” on all endpoints but it doesn’t help. if native_rtp not work for some reason I have oneway audio. how can I fix this? if I add mix_monitor it works, but it’s not a right way to fix this issues. Asterisk 13.2.0-- _ -- 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] TLS connect() error when calling udp to tls
Stuck with TLS transport, I have 2 phones (both in local network for tests) one connected with up second with tls when I calling TLS to UPD everything is fine, but when UDP calls TLS I getting an error ERROR[44230]: pjsip:0 : tlsc0x7f143012 TLS connect() error: Connection refused [code=120111] pjsip log: -- Called PJSIP/601/sip:601@192.168.1.55:5075;transport=tls <--- Transmitting SIP request (1052 bytes) to TLS:192.168.1.55:5075 ---> INVITE sip:601@192.168.1.55:5075;transport=tls SIP/2.0 Via: SIP/2.0/TLS 192.168.1.4:60410;rport;branch=z9hG4bKPj904eb4dc-b086-40c7-8ff1-4ddbaeea17f6;alias From: "" ;tag=5fc67f0a-2b96-469a-9d57-7b1d0ea8c1d3 To: Contact: Call-ID: 5ca66561-5755-4f1f-a951-2e6970aeeeda CSeq: 28062 INVITE Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, REFER, MESSAGE Supported: 100rel, timer, replaces, norefersub Session-Expires: 1800 Min-SE: 90 Max-Forwards: 70 User-Agent: PBXe 1.4.0 Content-Type: application/sdp Content-Length: 342 v=0 o=- 772596305 772596305 IN IP4 192.168.1.4 s=Asterisk c=IN IP4 192.168.1.4 t=0 0 m=audio 14476 RTP/SAVP 0 8 101 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:Ojz7o69EOsPsdsRTgNO/wtRWPsrWc2NSnOidNcqh 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=maxptime:150 a=sendrecv both phones SPA502, force_rport disabled for tls phone, here is my transports: [tls] type=transport ca_list_file=/pbx/keys/asterisk.pem cert_file=/pbx/keys/asterisk.crt priv_key_file=/pbx/keys/asterisk.key method=sslv23 protocol=tls bind=192.168.1.4:5061 external_media_address=8.8.8.8:5061 external_signaling_address=8.8.8.8:5061 [udp] type=transport protocol=udp bind=192.168.1.4 local_net=192.168.1.0/24 external_media_address=8.8.8.8 external_signaling_address=8.8.8.8 -- _ -- 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] Cannot configure PJSIP TLS
by removed line ca_list_file=/pbx/keys/ca.key ERROR[3301]: pjsip:0 : ssl0x7fc8e40f8 Error loading CA list file '/pbx/keys/ca.key gone. But still cannot handle SRTP, phone says 488 error if I set media_encryption=sdes on an endpoint, how do I check if srtp actually work on asterisk? > On 03 Mar 2015, at 20:14, Nick Awesome wrote: > > Hey guys,tried to make tls work with pjsip on asterisk 13.2.0 > > have compiled pjsip with ssl, > > added transport > > [tls] > type=transport > cert_file=/pbx/keys/server.crt > ca_list_file=/pbx/keys/ca.key > priv_key_file=/pbx/keys/server.key > protocol=tls > bind=192.168.1.4:5061 > local_net=192.168.1.0/24 > external_media_address=77.77.77.77 > external_signaling_address=77.77.77.77 > > have configured Grandstream GXP1400 to use tis and srtp, server.crt and > server.key uploaded to phone > > ubuntu*CLI> pjsip show transports > Transport: tls tls 0 0 192.168.1.4:5061 > > so transport exist, have set endpoint transport to tls, > > but for some reason phone getting timeout 408. tried from local network and > behind the nat, nothing. -- _ -- 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] Cannot configure PJSIP TLS
Hey guys,tried to make tls work with pjsip on asterisk 13.2.0 have compiled pjsip with ssl, added transport [tls] type=transport cert_file=/pbx/keys/server.crt ca_list_file=/pbx/keys/ca.key priv_key_file=/pbx/keys/server.key protocol=tls bind=192.168.1.4:5061 local_net=192.168.1.0/24 external_media_address=77.77.77.77 external_signaling_address=77.77.77.77 have configured Grandstream GXP1400 to use tis and srtp, server.crt and server.key uploaded to phone ubuntu*CLI> pjsip show transports Transport: tls tls 0 0 192.168.1.4:5061 so transport exist, have set endpoint transport to tls, but for some reason phone getting timeout 408. tried from local network and behind the nat, nothing. -- _ -- 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] having trouble to register cisco 7975 with pjsip
success! just replaced MeetMe to Bridge in softkey.xml and conf works now with the latest fw! On Feb 26, 2015, at 9:00 AM, Nick Awesome wrote: > > I have not working 3way conference, when I trying to connect second call, > phone says “unable to set up conference” > and sending some cisco xml data to asterisk which cannot be handled, thats > the problem, > -- _ -- 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] having trouble to register cisco 7975 with pjsip
another issues with cisco 7975 I have phone registered on asterisk have 2 different issues on different versions of firmware, on 9-4-2-1S I have not working 3way conference, when I trying to connect second call, phone says “unable to set up conference” and sending some cisco xml data to asterisk which cannot be handled, thats the problem, I know on firmware 8-5-4 3way conference works just fine 3cx phone system so must be same with asterisk, but with asterisk when I do ANY call from cisco phone with fw 8-5-4 cisco hangup call after channels connect, debug <--- Received SIP request (1003 bytes) from UDP:192.168.1.61:49163 ---> INVITE sip:*777@192.168.1.4;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.1.61:5060;branch=z9hG4bKa67a2ab7 From: "111" ;tag=0c8525a689610012e85fd91b-ee689f06 To: Call-ID: 0c8525a6-89610012-53a7b585-a4dc85a0@192.168.1.61 Max-Forwards: 70 Date: Thu, 26 Feb 2015 05:52:42 GMT CSeq: 101 INVITE User-Agent: Cisco-CP7975G/8.5.3 Contact: Expires: 180 Accept: application/sdp Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE Allow-Events: kpml,dialog Content-Length: 322 Content-Type: application/sdp Content-Disposition: session;handling=optional v=0 o=Cisco-SIPUA 626 0 IN IP4 192.168.1.61 s=SIP Call t=0 0 m=audio 30354 RTP/AVP 0 8 18 116 101 c=IN IP4 192.168.1.61 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:116 iLBC/8000 a=fmtp:116 mode=20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv <--- Transmitting SIP response (485 bytes) to UDP:192.168.1.61:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.61:5060;received=192.168.1.61;branch=z9hG4bKa67a2ab7 Call-ID: 0c8525a6-89610012-53a7b585-a4dc85a0@192.168.1.61 From: "111" ;tag=0c8525a689610012e85fd91b-ee689f06 To: ;tag=z9hG4bKa67a2ab7 CSeq: 101 INVITE WWW-Authenticate: Digest realm="asterisk",nonce="1424929962/9af5af19e633c82d2a9e17ec97afb72b",opaque="2776507e426bda2b",algorithm=md5,qop="auth" Content-Length: 0 <--- Received SIP request (368 bytes) from UDP:192.168.1.61:49174 ---> ACK sip:*777@192.168.1.4;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.1.61:5060;branch=z9hG4bKa67a2ab7 From: "111" ;tag=0c8525a689610012e85fd91b-ee689f06 To: ;tag=z9hG4bKa67a2ab7 Call-ID: 0c8525a6-89610012-53a7b585-a4dc85a0@192.168.1.61 Date: Thu, 26 Feb 2015 05:52:42 GMT CSeq: 101 ACK Content-Length: 0 <--- Received SIP request (1271 bytes) from UDP:192.168.1.61:49163 ---> INVITE sip:*777@192.168.1.4;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.1.61:5060;branch=z9hG4bK4affb043 From: "111" ;tag=0c8525a689610012e85fd91b-ee689f06 To: Call-ID: 0c8525a6-89610012-53a7b585-a4dc85a0@192.168.1.61 Max-Forwards: 70 Date: Thu, 26 Feb 2015 05:52:42 GMT CSeq: 102 INVITE User-Agent: Cisco-CP7975G/8.5.3 Contact: Authorization: Digest username="111",realm="asterisk",uri="sip:*777@192.168.1.4;user=phone",response="8b90970d8fc724893e876263ce8c2cd3",nonce="1424929962/9af5af19e633c82d2a9e17ec97afb72b",opaque="2776507e426bda2b",cnonce="945bf4a1",qop=auth,nc=0001,algorithm=md5 Expires: 180 Accept: application/sdp Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE Allow-Events: kpml,dialog Content-Length: 322 Content-Type: application/sdp Content-Disposition: session;handling=optional v=0 o=Cisco-SIPUA 626 0 IN IP4 192.168.1.61 s=SIP Call t=0 0 m=audio 30354 RTP/AVP 0 8 18 116 101 c=IN IP4 192.168.1.61 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:116 iLBC/8000 a=fmtp:116 mode=20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv <--- Transmitting SIP response (312 bytes) to UDP:192.168.1.61:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.61:5060;received=192.168.1.61;branch=z9hG4bK4affb043 Call-ID: 0c8525a6-89610012-53a7b585-a4dc85a0@192.168.1.61 From: "111" ;tag=0c8525a689610012e85fd91b-ee689f06 To: CSeq: 102 INVITE Content-Length: 0 <--- Transmitting SIP response (738 bytes) to UDP:192.168.1.61:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.61:5060;received=192.168.1.61;branch=z9hG4bK4affb043 Call-ID: 0c8525a6-89610012-53a7b585-a4dc85a0@192.168.1.61 From: "111" ;tag=0c8525a689610012e85fd91b-ee689f06 To: ;tag=916a8d96-8a85-4474-b404-e30615c6c963 CSeq: 102 INVITE Contact: Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, REFER, MESSAGE Supported: 100rel, timer, replaces, norefersub Content-Type: application/sdp Content-Length: 163 v=0 o=- 626 2 IN IP4 192.168.1.4 s=Asterisk c=IN IP4 192.168.1.4 t=0 0 m=audio 10474 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=ptime:20 a=maxptime:150 a=sendrecv <--- Received SIP request (697 bytes) from UDP:192.168.1.61:49163 ---> ACK sip:192.168.1.4:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.61:5060;branch=z9hG4bK22ad7045 From: "111" ;tag=0c8525a689610012e85fd91b-ee689f06 To: ;tag=916a8d96-8a85-4474-b404-e30615c6c963 Call-ID: 0c8525a6-89610012-53a7b585-a4dc85a0@192.168.1.61 Max-Forwards: 70 Date: Thu, 26 Feb
Re: [asterisk-users] having trouble to register cisco 7975 with pjsip
Oh god it works ! to switch cisco to upd I used config: 2 with udp it works well, thanks for your help :) > On 24 Feb 2015, at 17:02, Joshua Colp wrote: > > If you use UDP with force_rport=no it'll work. > If you use TCP then set rewrite_contact=yes so it'll reuse the established > TCP connection. -- _ -- 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] having trouble to register cisco 7975 with pjsip
Ok after I added tcp transport and disable force_rport phone get registered, but still have issues with calls, when I call from cisco from, it work except hangup. when I call to cisco phone asterisk return congested debug of call <--- Transmitting SIP request (952 bytes) to TCP:192.168.1.61:51179 ---> INVITE sip:111@192.168.1.61:51179;transport=tcp SIP/2.0 Via: SIP/2.0/TCP 192.168.1.4:55246;rport;branch=z9hG4bKPjcb9ec9ba-0c77-4530-a3b7-44209357f3a0;alias From: ;tag=abebd75c-501a-4b4f-ad69-ee98175b8dbd To: Contact: Call-ID: bb515935-7292-47b4-890d-6f82eb335815 CSeq: 25333 INVITE Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, REFER, MESSAGE Supported: 100rel, timer, replaces, norefersub Session-Expires: 1800 Min-SE: 90 Content-Type: application/sdp Content-Length: 283 v=0 o=- 1231372975 1231372975 IN IP4 192.168.1.4 s=Asterisk c=IN IP4 192.168.1.4 t=0 0 m=audio 17856 RTP/AVP 9 0 8 101 a=rtpmap:9 G722/8000 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=maxptime:150 a=sendrecv [Feb 24 05:47:01] WARNING[16179]: pjsip:0 : tsx0x7f1aa0157 Failed to send Request msg INVITE/cseq=12216 (tdta0x7f1aa00e41c0)! err=120111 (Connection refused) [Feb 24 05:47:01] ERROR[16179]: pjsip:0 :tcpc0x7f1aa01c TCP connect() error: Connection refused [code=120111] [Feb 24 05:47:01] WARNING[16179]: pjsip:0 : tsx0x7f1aa01c3 Failed to send Request msg INVITE/cseq=25333 (tdta0x7f1aa00ad810)! err=120111 (Connection refused) > On 24 Feb 2015, at 15:05, Joshua Colp wrote: > > Nick Awesome wrote: >> Hay guys, got trouble with registration with cisco 7975 > > The "force_rport" option is incompatible with Cisco, it needs to be > explicitly set to no in the endpoint. > > Cheers, > > -- > 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 -- > 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] having trouble to register cisco 7975 with pjsip
Hay guys, got trouble with registration with cisco 7975 Here is the debug : <--- Received SIP request (576 bytes) from UDP:192.168.1.61:49533 ---> REGISTER sip:192.168.1.4 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.61:5060;branch=z9hG4bK35076381 From: ;tag=0c8525a68961001f44d503e2-d9359bd3 To: Call-ID: 0c8525a6-89610004-b972d038-5864c98e@192.168.1.61 Max-Forwards: 70 Date: Tue, 24 Feb 2015 07:13:42 GMT CSeq: 110 REGISTER User-Agent: Cisco-CP7975G/8.5.3 Contact: ;+sip.instance="";+u.sip!model.ccm.cisco.com="437" Supported: (null),X-cisco-xsi-7.0.1 Content-Length: 0 Expires: 3600 <--- Transmitting SIP response (481 bytes) to UDP:192.168.1.61:49531 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.61:5060;rport=49531;received=192.168.1.61;branch=z9hG4bKd16b1eb7 Call-ID: 0c8525a6-89610002-845d0080-f3559596@192.168.1.61 From: ;tag=0c8525a68961001d53245ebc-a1b56549 To: ;tag=z9hG4bKd16b1eb7 CSeq: 110 REGISTER WWW-Authenticate: Digest realm="asterisk",nonce="1424762038/41d5874af9ea9408c257949c309c8aa0",opaque="7f15d8c2312c7b0d",algorithm=md5,qop="auth" Content-Length: 0 username and password are correct, this phone was working with 3CX just fine but won’t work with asterisk for some reason. ( any idea what may cause the problem?-- _ -- 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] Queue PJSIP, not all contacts rings
Works, thank you!
> On Feb 23, 2015, at 7:11 PM, Joshua Colp wrote:
>
> Nick Awesome wrote:
>> Hay guys, have question.
>>
>> When I do regular dial I use
>> $this->AGI->get_fullvariable('${PJSIP_DIAL_CONTACTS('.$callObj.')}',false,true);
>>
>> to get all contacts of current endpoint and so I dial to all phones
>> at once,
>>
>> but if I exec QUEUE, I have just one phone rings, seems like it take
>> first one as Dial app by default, is there way to fix this?
>
> There is no way to directly do this. The best option is to use a Local
> channel into the dialplan which dials instead. Once answered everything
> should fall into place.
>
> --
> 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 --
> 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 --
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asterisk-users mailing list
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[asterisk-users] Queue PJSIP, not all contacts rings
Hay guys, have question.
When I do regular dial I use
$this->AGI->get_fullvariable('${PJSIP_DIAL_CONTACTS('.$callObj.')}',false,true);
to get all contacts of current endpoint and so I dial to all phones at once,
but if I exec QUEUE, I have just one phone rings, seems like it take first one
as Dial app by default, is there way to fix this?
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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Re: [asterisk-users] Sent ami event from AGI?
Works! how I miss that… Thanks. On 02 Oct 2014, at 17:05, Scott Griepentrog wrote: > You can use the AGI command EXEC to execute a dialplan application, and the > application UserEvent can be used to generate custom events that AMI clients > can receive. > > https://wiki.asterisk.org/wiki/display/AST/AGICommand_exec > > https://wiki.asterisk.org/wiki/display/AST/Application_UserEvent > > > > On Thu, Oct 2, 2014 at 4:02 AM, Ilya Awesome wrote: > hello, is there way to send event to all ami clients from AGI script? > > Sent from my iPhone > > -- > _ > -- 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 > > > > -- > > Scott Griepentrog > Digium, Inc · Software Developer > 445 Jan Davis Drive NW · Huntsville, AL 35806 · US > direct/fax: +1 256 428 6239 · mobile: +1 317 507 4029 > Check us out at: http://digium.com · http://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] PJSIP issues with handling incoming calls
Ok, thanks for an answer. That solution works.
On 02 Sep 2014, at 22:36, Rainer Piper wrote:
> contact_user in pjsip.conf has to point to the filter or to an agi in the
> extentions.conf
> like:
>
> pjsip.conf
> contact_user=blablabla
>
> extensions.conf
> exten => blablabla, 1,NoOp(*** ${PJSIP_HEADER(read,To)} *** ${CALLERID(num)}
> ***)
>
>
> Am 02.09.2014 um 20:11 schrieb Rainer Piper:
>> contact_user can be anything and calling an agi is no problem
>>
>>
>> Am 02.09.2014 um 19:49 schrieb Nick Awesome:
>>> Okay, contact_user seems like do the job. Thanks
>>> is contact_user can be anything, or it should be same as username ?
>>> I would like to use contact_user for transmitting trunk name into agi script
>>>
>>> On Sep 2, 2014, at 7:04 PM, Rainer Piper wrote:
>>>
>>>> I use in pjsip.conf
>>>> [sipgate1]
>>>> type=registration
>>>> transport=transport-udp
>>>> outbound_auth=sipgate1_auth
>>>> server_uri=sip:sipgate.de
>>>> client_uri=sip:555123...@sipgate.de
>>>> contact_user=sipgatefilter ; goto the filter in extensions.conf
>>>> retry_interval=60
>>>> forbidden_retry_interval=600
>>>> expiration=3600
>>>>
>>>> extensions.conf ; i'm cutting the dialed number out of the invite Header
>>>> and goto/jump to the extensions
>>>> ; incoming VOIP 9716716x SIPGATE
>>>> exten => sipgatefilter,1,NoOp(*** ${PJSIP_HEADER(read,To)} ***
>>>> ${CALLERID(num)} ***)
>>>> same => n,Set(gotoadr=${CUT(PJSIP_HEADER(read,To),@,1)})
>>>> same => n,NoOp( 49${gotoadr:-11} )
>>>> same => n,Goto(49${gotoadr:-11},1)
>>>>
>>>> ; the filter is jumping to the extensions ...
>>>>
>>>> ; incoming VOIP 97167160 SIPGATE -> MENU
>>>> exten =>
>>>> 4922897167160,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.de&PJSIP/7000&PJSIP/7004&PJSIP/7003&PJSIP/7005,,r)
>>>> ; incoming VOIP 97167161 SIPGATE
>>>> exten =>
>>>> 4922897167161,1,Dial(PJSIP/7000&PJSIP/7001&PJSIP/7003&PJSIP/7004,,r)
>>>> ; incoming VOIP 97167162 SIPGATE ECHO TEST
>>>> exten =>
>>>> 4922897167162,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.de&PJSIP/7000,,r)
>>>> ; incoming VOIP 97167163 SIPGATE
>>>> exten =>
>>>> 4922897167163,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.de&PJSIP/7000,,r)
>>>> ; incoming VOIP 97167164 SIPGATE
>>>> exten =>
>>>> 4922897167164,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.de&PJSIP/7000,,r)
>>>> ; incoming VOIP 97167165 SIPGATE
>>>> exten =>
>>>> 4922897167165,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.de&PJSIP/7000,,r)
>>>> ; incncoming VOIP 97167166 Mailbox
>>>> exten =>
>>>> 4922897167166,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.de&PJSIP/7000,,r)
>>>> ; incoming VOIP 97167167 CONF. 1
>>>> exten =>
>>>> 4922897167167,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.de&PJSIP/7000,,r)
>>>> ; incoming VOIP 97167168 CONF. 2
>>>> ;exten =>
>>>> 4922897167168,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.de&PJSIP/7000,,r)
>>>> exten => 4922897167168,1,Answer
>>>> same => n,echo()
>>>> same => n,Hangup()
>>>> ; incoming VOIP 97167169 FAX
>>>> ;exten =>
>>>> 4922897167169,1,Dial(PJSIP/${EXTEN}@sip.soho-piper.de&PJSIP/7000,,r)
>>>>
>>>>
>>>> Regards
>>>> Rainer
>>>>
>>>> Am 02.09.2014 um 15:08 schrieb Joshua Colp:
>>>>> Nick Awesome wrote:
>>>>>> register => 73432260005:pass@10001
>>>>>> register => 73432260050:pass@10002
>>>>>>
>>>>>> [10001]
>>>>>> type=peer
>>>>>> host=80.75.132.66
>>>>>> context=dialmap
>>>>>> [10002]
>>>>>> type=peer
>>>>>> host=80.75.132.66
>>>>>> context=dialmap
>>>>>
>>>>> Can you provide a sip debug of calls to both of these? I'm confused how
>>>>> that... works...
>>>>>
>>>>
>>>>
>>>> --
>>>> Rainer Piper
>>>> Integration engineer
>>>> Koesli
Re: [asterisk-users] PJSIP issues with handling incoming calls
Okay, contact_user seems like do the job. Thanks
is contact_user can be anything, or it should be same as username ?
I would like to use contact_user for transmitting trunk name into agi script
On Sep 2, 2014, at 7:04 PM, Rainer Piper wrote:
> I use in pjsip.conf
> [sipgate1]
> type=registration
> transport=transport-udp
> outbound_auth=sipgate1_auth
> server_uri=sip:sipgate.de
> client_uri=sip:555123...@sipgate.de
> contact_user=sipgatefilter ; goto the filter in extensions.conf
> retry_interval=60
> forbidden_retry_interval=600
> expiration=3600
>
> extensions.conf ; i'm cutting the dialed number out of the invite Header and
> goto/jump to the extensions
> ; incoming VOIP 9716716x SIPGATE
> exten => sipgatefilter,1,NoOp(*** ${PJSIP_HEADER(read,To)} ***
> ${CALLERID(num)} ***)
> same => n,Set(gotoadr=${CUT(PJSIP_HEADER(read,To),@,1)})
> same => n,NoOp( 49${gotoadr:-11} )
> same => n,Goto(49${gotoadr:-11},1)
>
> ; the filter is jumping to the extensions ...
>
> ; incoming VOIP 97167160 SIPGATE -> MENU
> exten =>
> 4922897167160,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.de&PJSIP/7000&PJSIP/7004&PJSIP/7003&PJSIP/7005,,r)
> ; incoming VOIP 97167161 SIPGATE
> exten => 4922897167161,1,Dial(PJSIP/7000&PJSIP/7001&PJSIP/7003&PJSIP/7004,,r)
> ; incoming VOIP 97167162 SIPGATE ECHO TEST
> exten =>
> 4922897167162,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.de&PJSIP/7000,,r)
> ; incoming VOIP 97167163 SIPGATE
> exten =>
> 4922897167163,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.de&PJSIP/7000,,r)
> ; incoming VOIP 97167164 SIPGATE
> exten =>
> 4922897167164,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.de&PJSIP/7000,,r)
> ; incoming VOIP 97167165 SIPGATE
> exten =>
> 4922897167165,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.de&PJSIP/7000,,r)
> ; incncoming VOIP 97167166 Mailbox
> exten =>
> 4922897167166,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.de&PJSIP/7000,,r)
> ; incoming VOIP 97167167 CONF. 1
> exten =>
> 4922897167167,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.de&PJSIP/7000,,r)
> ; incoming VOIP 97167168 CONF. 2
> ;exten =>
> 4922897167168,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.de&PJSIP/7000,,r)
> exten => 4922897167168,1,Answer
> same => n,echo()
> same => n,Hangup()
> ; incoming VOIP 97167169 FAX
> ;exten => 4922897167169,1,Dial(PJSIP/${EXTEN}@sip.soho-piper.de&PJSIP/7000,,r)
>
>
> Regards
> Rainer
>
> Am 02.09.2014 um 15:08 schrieb Joshua Colp:
>> Nick Awesome wrote:
>>> register => 73432260005:pass@10001
>>> register => 73432260050:pass@10002
>>>
>>> [10001]
>>> type=peer
>>> host=80.75.132.66
>>> context=dialmap
>>> [10002]
>>> type=peer
>>> host=80.75.132.66
>>> context=dialmap
>>
>> Can you provide a sip debug of calls to both of these? I'm confused how
>> that... works...
>>
>
>
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> Phone: +49 228 97167161
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Re: [asterisk-users] PJSIP issues with handling incoming calls
register => 73432260005:pass@10001 register => 73432260050:pass@10002 [10001] type=peer host=80.75.132.66 context=dialmap [10002] type=peer host=80.75.132.66 context=dialmap so now in context dialmap (agi application) AGI->agi_channel is 'SIP/10001-0005’ parsing 10001 and checking db for matches, in db I have table with all my trunks information On 02 Sep 2014, at 15:49, Joshua Colp wrote: > Nick Awesome wrote: >> Tried doing that, but >> >> first: AGI->exten is ’s’ for some reason. and second its not >> practical, for example if 80.75.132.66 wound like to register on my * >> server - it will not work because I already using that IP with >> different endpoint >> >> well, its critical trouble for me, coming back to chat_sip :| > > How will you do this in chan_sip? The behavior between the two is the same, > despite the configuration being different. > > -- > 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 -- > 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] PJSIP issues with handling incoming calls
Tried doing that, but
first: AGI->exten is ’s’ for some reason.
and second its not practical, for example if 80.75.132.66 wound like to
register on my * server - it will not work because I already using that IP with
different endpoint
well, its critical trouble for me, coming back to chat_sip :|
On 02 Sep 2014, at 15:32, A J Stiles wrote:
> On Tuesday 02 Sep 2014, Nick Awesome wrote:
>> Hello guys.
>>
>> Have 2 external numbers that required registration on provider server,
>>
>> trunk1: 73432260005@80.75.132.66
>> trunk2: 73432260050@80.75.132.66
>>
>> Thing is I can’t figure out how to route them to different IVRs
>>
>> by default Asterisk can’t match endpoint
>>
>> Request from '' failed for
>> '80.75.132.66:5060' (callid: 50e9132765782741404408k2469rmwp) - No
>> matching endpoint found
>>
>> Can’t set identify by IP because they got the same ip.
>>
>> Is there way to configure asterisk so incoming calls from same IP but
>> different ID will use different contexts?
>
> Can't you send them both to the same context initially; but once you are
> there, match the outside number (which can be found in ${EXTEN} if it is the
> number that was dialled from their end, or ${CALLERID(num)} if it is the
> number they are calling from) within that context and use a GoToIf() to send
> calls from trunk 2 to the correct context?
>
> --
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>
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> list, change address to asterisk1list at earthshod dot co dot uk .
>
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Re: [asterisk-users] PJSIP issues with handling incoming calls
Thats because I call from one to other here’s logs where I call from mobile <--- Received SIP request (469 bytes) from UDP:80.75.132.66:5060 ---> ACK sip:s@pbx_ip_address:57408;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 80.75.132.66:5060;branch=z9hG4bK-524287-1-NGJlZjMxNThhYjI2YjI3Y2EyODE0MThhMTVkNjY0ZTA.--5c6da819e6300b26;rport Max-Forwards: 70 To: ;tag=z9hG4bK-524287-1-NGJlZjMxNThhYjI2YjI3Y2EyODE0MThhMTVkNjY0ZTA.--5c6da819e6300b26 From: ;tag=7ozmpvsvqs26kcor.o Call-ID: 18e2786560719216837824k41099rmwp CSeq: 586 ACK Content-Length: 0 <--- Received SIP request (469 bytes) from UDP:80.75.132.66:5060 ---> ACK sip:s@pbx_ip_address:57408;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 80.75.132.66:5060;branch=z9hG4bK-524287-1-YzhmMDE2NzE1YjRhOTM4NjQ1MjMxMmMyMmM0MWFiZTE.--be7c48325cdef400;rport Max-Forwards: 70 To: ;tag=z9hG4bK-524287-1-YzhmMDE2NzE1YjRhOTM4NjQ1MjMxMmMyMmM0MWFiZTE.--be7c48325cdef400 From: ;tag=yddmzvcoi3waw24e.o Call-ID: 22e7064301970213226722k41100rmwp CSeq: 588 ACK Content-Length: 0 On 02 Sep 2014, at 15:01, Joshua Colp wrote: > Nick Awesome wrote: >> Hello guys. > > Kia ora, > >> Have 2 external numbers that required registration on provider server, >> >> trunk1: 734322600*05*@80.75.132.66 >> trunk2: 734322600*50*@80.75.132.66 >> >> Thing is I can’t figure out how to route them to different IVRs >> >> by default Asterisk can’t match endpoint >> >> Request from '' failed for >> '80.75.132.66:5060' (callid: 50e9132765782741404408k2469rmwp) - No >> matching endpoint found >> >> Can’t set /identify /by IP because they got the same ip. >> >> Is there way to configure asterisk so incoming calls from same IP but >> different ID will use different contexts? > > If the From header contains the destination number (as it seems to based on > your above log message and config) you can create two different endpoints and > match based on the user portion of the From header. > > [734322600*05*] > type=endpoint > context=did-1 > disallow=all > allow=ulaw > > [734322600*50*] > type=endpoint > context=did-2 > disallow=all > allow=ulaw > > If this is not correct then you can only match once based on the source IP > address currently. > > Cheers, > > -- > 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 -- > 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] PJSIP issues with handling incoming calls
So there is no way to do that with pjsip? On 02 Sep 2014, at 11:35, Administrator TOOTAI wrote: > Le 02/09/2014 08:47, Nick Awesome a écrit : >> Hello guys. > > Hi > >> >> Have 2 external numbers that required registration on provider server, >> >> trunk1: 734322600*05*@80.75.132.66 >> trunk2: 734322600*50*@80.75.132.66 >> >> Thing is I can’t figure out how to route them to different IVRs >> >> by default Asterisk can’t match endpoint >> >> Request from '' failed for >> '80.75.132.66:5060' (callid: 50e9132765782741404408k2469rmwp) - No >> matching endpoint found >> >> Can’t set /identify /by IP because they got the same ip. >> >> Is there way to configure asterisk so incoming calls from same IP but >> different ID will use different contexts? > > You have to register to the gateway with each account user and password like > > sip.conf > > register = 734322600*05*:password1@myProvider/734322600*05* > register = 734322600*50*:password2@myProvider/734322600*50* > > [myProvider] > type=peer > host=80.75.132.66 > context=from-myProvider > ... > > extensions.conf > > [from-myProvider] > exten = 734322600*05*,1,NoOp(Incoming call to 734322600*05*) > ... > > exten = 734322600*50*,1,NoOp(Incoming call to 734322600*50*) > ... > > -- > _ > -- 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] PJSIP issues with handling incoming calls
Hello guys. Have 2 external numbers that required registration on provider server, trunk1: 73432260005@80.75.132.66 trunk2: 73432260050@80.75.132.66 Thing is I can’t figure out how to route them to different IVRs by default Asterisk can’t match endpoint Request from '' failed for '80.75.132.66:5060' (callid: 50e9132765782741404408k2469rmwp) - No matching endpoint found Can’t set identify by IP because they got the same ip. Is there way to configure asterisk so incoming calls from same IP but different ID will use different contexts?-- _ -- 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] Hold ,UnHold Via AMI
Probably you should use “Action: Park" example: Action: Park Channel: SIP/1000-0003 Channel2: SIP/1000-0004 On 21 Jul 2014, at 17:00, mahdieh saeed wrote: > Hi, > I want to write API for doing some actions. I want to write function for hold > special call via AMI.But I can not find any action for this purpose. > Is there any action for holding special channel? > > Regards, > Mahdieh Saeed > -- > _ > -- 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] Asterisk 14.4.0 MeetMe crash
Hi, after update on 12.4.0 asterisk crashes on MeetMe ending on 12.3.2 it worked well. Is some one else have this issues? should someone open a ticket? -- _ -- 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] Transfer call question
Hello guys, I have trunk “1", Internal num “99" and MeetMe “1010" now I calling 99 -> 89264959635 via 1 /pbx/agi.php: [agi_channel] => PJSIP/99-0012 /pbx/agi.php: [agi_callerid] => 99 /pbx/agi.php: [agi_calleridname] => 99 /pbx/agi.php: [agi_context] => dialmap /pbx/agi.php: [agi_extension] => 89264959635 then I would like to direct transfer this call to 1010 and when I do that from my phone I getting this agi_request in AGI: /pbx/agi.php: [agi_channel] => PJSIP/1-0013 /pbx/agi.php: [agi_callerid] => 89264959635 /pbx/agi.php: [agi_calleridname] => unknown /pbx/agi.php: [agi_context] => dialmap /pbx/agi.php: [agi_extension] => 1010 There is no information who is transferring that call, so AGI thinks that it is inbound call and hangup it because in my case "external 89264959635 to internal 1010 is denied". is there way do determine that call was transfered from 99 so I can use route table of abonent 99 to connect the call properly? -- _ -- 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] PJSIP outbound register and inbound calls
oh.. its simple. "[res_pjsip_endpoint_identifier_ip]" should be before "identify=realtime,ps_endpoint_id_ips”, not "[res_pjsip]” Thanks all for help :) On 17 Jul 2014, at 11:05, Nick Awesome wrote: > New information, as I said I’m using realtime, > thats the problem! > > I just tested using static config file and it is working perfect. > after some research I figured out that problem with “ps_endpoint_id_ips" for > some reason asterisk ignoring matches in this table, > > I have string in sorcery.conf > > identify = realtime,ps_endpoint_id_ips > > also have string in extconfig.conf > > ps_endpoint_id_ips => odbc,asterisk,pbx_endpoint_id_ips > > and ofc I have table > > CREATE TABLE `pbx_endpoint_id_ips` ( > `id` varchar(40) NOT NULL, > `endpoint` varchar(40) DEFAULT NULL, > `match` varchar(80) DEFAULT NULL, > UNIQUE KEY `id` (`id`), > KEY `ps_endpoint_id_ips_id` (`id`) > ) ENGINE=InnoDB DEFAULT CHARSET=latin1; > > with entry > > 10001 | 10001 | 85.195.98.178 > > but thats just didn’t works( > > is this a bug and should I open ticket ? > > On 16 Jul 2014, at 21:13, Nick Awesome wrote: > >> Ok there is my test account from sipiko.net >> >> username: cb5069 >> password: sqv664yqtp >> domain: callme.sipiko.net >> >> its using username/password authentication. >> because its just website widget I need only inbound calls from this peer, >> test call can be done from url: >> http://callme.sipiko.net/callme.php?id=5069&call_id=210&tunnel=yes >> >> on my side I have an asterisk 12 using pjsip >> >> Have configured IVR with number 5000 on context "dialmap", so I need forward >> all calls from this provider to number 5000 over "dialmap" context >> >> help if you can please:) >> >> On Jul 16, 2014, at 8:53 PM, Joshua Colp wrote: >> >>> Nick Awesome wrote: >>>> I thought that >>>>>> type=identify >>>> will match an IP address and accept it, >>>> >>>> well, in my example I can control both sides and able to configure it >>>> without registration. in real life I have a provider that requires >>>> username/password authentication >>>> >>>> provider gives me - Username - Password - DomainName >>> >>> They may require it for *outgoing* calls to them but for incoming I >>> highly doubt they'd want you to authenticate them. It's usually always >>> IP authentication. >>> >>>> I have configure it like I showed before and have exactly the same >>>> notice >>>> >>>> [Jul 16 20:32:23] NOTICE[21926]: res_pjsip/pjsip_distributor.c:246 >>>> log_unidentified_request: Request from >>>> '"cb5069"' failed for >>>> '85.195.98.178:5060' (callid: >>>> 173995aa2e25283807700d65055c9214@85.195.98.178) - No matching >>>> endpoint found 85.195.98.178 is an operator, >>>> >>>> so what I should add to my config to be able accept calls from >>>> Registered peer ? >>> >>> The PJSIP functionality does not currently allow using the dynamic IP of a >>> registration to match an incoming call. You either have to explicitly use >>> the identify section or match as I previously described. >>> >>> Without further details of your setup (IP addresses, who are calling who) >>> and how you want it to work I can't answer. >>> >>> -- >>> 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 -- >>> 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 -- _ -- 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] PJSIP outbound register and inbound calls
New information, as I said I’m using realtime, thats the problem! I just tested using static config file and it is working perfect. after some research I figured out that problem with “ps_endpoint_id_ips" for some reason asterisk ignoring matches in this table, I have string in sorcery.conf identify = realtime,ps_endpoint_id_ips also have string in extconfig.conf ps_endpoint_id_ips => odbc,asterisk,pbx_endpoint_id_ips and ofc I have table CREATE TABLE `pbx_endpoint_id_ips` ( `id` varchar(40) NOT NULL, `endpoint` varchar(40) DEFAULT NULL, `match` varchar(80) DEFAULT NULL, UNIQUE KEY `id` (`id`), KEY `ps_endpoint_id_ips_id` (`id`) ) ENGINE=InnoDB DEFAULT CHARSET=latin1; with entry 10001 | 10001 | 85.195.98.178 but thats just didn’t works( is this a bug and should I open ticket ? On 16 Jul 2014, at 21:13, Nick Awesome wrote: > Ok there is my test account from sipiko.net > > username: cb5069 > password: sqv664yqtp > domain: callme.sipiko.net > > its using username/password authentication. > because its just website widget I need only inbound calls from this peer, > test call can be done from url: > http://callme.sipiko.net/callme.php?id=5069&call_id=210&tunnel=yes > > on my side I have an asterisk 12 using pjsip > > Have configured IVR with number 5000 on context "dialmap", so I need forward > all calls from this provider to number 5000 over "dialmap" context > > help if you can please:) > > On Jul 16, 2014, at 8:53 PM, Joshua Colp wrote: > >> Nick Awesome wrote: >>> I thought that >>>>> type=identify >>> will match an IP address and accept it, >>> >>> well, in my example I can control both sides and able to configure it >>> without registration. in real life I have a provider that requires >>> username/password authentication >>> >>> provider gives me - Username - Password - DomainName >> >> They may require it for *outgoing* calls to them but for incoming I >> highly doubt they'd want you to authenticate them. It's usually always >> IP authentication. >> >>> I have configure it like I showed before and have exactly the same >>> notice >>> >>> [Jul 16 20:32:23] NOTICE[21926]: res_pjsip/pjsip_distributor.c:246 >>> log_unidentified_request: Request from >>> '"cb5069"' failed for >>> '85.195.98.178:5060' (callid: >>> 173995aa2e25283807700d65055c9214@85.195.98.178) - No matching >>> endpoint found 85.195.98.178 is an operator, >>> >>> so what I should add to my config to be able accept calls from >>> Registered peer ? >> >> The PJSIP functionality does not currently allow using the dynamic IP of a >> registration to match an incoming call. You either have to explicitly use >> the identify section or match as I previously described. >> >> Without further details of your setup (IP addresses, who are calling who) >> and how you want it to work I can't answer. >> >> -- >> 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 -- >> 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] PJSIP outbound register and inbound calls
Ok there is my test account from sipiko.net username: cb5069 password: sqv664yqtp domain: callme.sipiko.net its using username/password authentication. because its just website widget I need only inbound calls from this peer, test call can be done from url: http://callme.sipiko.net/callme.php?id=5069&call_id=210&tunnel=yes on my side I have an asterisk 12 using pjsip Have configured IVR with number 5000 on context "dialmap", so I need forward all calls from this provider to number 5000 over "dialmap" context help if you can please:) On Jul 16, 2014, at 8:53 PM, Joshua Colp wrote: > Nick Awesome wrote: >> I thought that >>>> type=identify >> will match an IP address and accept it, >> >> well, in my example I can control both sides and able to configure it >> without registration. in real life I have a provider that requires >> username/password authentication >> >> provider gives me - Username - Password - DomainName > > They may require it for *outgoing* calls to them but for incoming I > highly doubt they'd want you to authenticate them. It's usually always > IP authentication. > >> I have configure it like I showed before and have exactly the same >> notice >> >> [Jul 16 20:32:23] NOTICE[21926]: res_pjsip/pjsip_distributor.c:246 >> log_unidentified_request: Request from >> '"cb5069"' failed for >> '85.195.98.178:5060' (callid: >> 173995aa2e25283807700d65055c9214@85.195.98.178) - No matching >> endpoint found 85.195.98.178 is an operator, >> >> so what I should add to my config to be able accept calls from >> Registered peer ? > > The PJSIP functionality does not currently allow using the dynamic IP of a > registration to match an incoming call. You either have to explicitly use the > identify section or match as I previously described. > > Without further details of your setup (IP addresses, who are calling who) and > how you want it to work I can't answer. > > -- > 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 -- > 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] PJSIP outbound register and inbound calls
I thought that >> type=identify will match an IP address and accept it, well, in my example I can control both sides and able to configure it without registration. in real life I have a provider that requires username/password authentication provider gives me - Username - Password - DomainName I have configure it like I showed before and have exactly the same notice [Jul 16 20:32:23] NOTICE[21926]: res_pjsip/pjsip_distributor.c:246 log_unidentified_request: Request from '"cb5069" ' failed for '85.195.98.178:5060' (callid: 173995aa2e25283807700d65055c9214@85.195.98.178) - No matching endpoint found 85.195.98.178 is an operator, so what I should add to my config to be able accept calls from Registered peer ? On Jul 16, 2014, at 7:55 PM, Joshua Colp wrote: > Nick Awesome wrote: >> Hi all, In my case I using realtime, here is how it looks in plant >> >> [10001] type=registration transport=upd_static outbound_auth=10001 >> server_uri=sip:600@192.168.1.1:5060 >> client_uri=sip:600@192.168.1.4:5060 [10001] type=auth >> auth_type=userpass password=600 username=600 [10001] type=aor >> contact=sip:192.168.1.4:5060 [10001] type=endpoint >> transport=upd_static context=dialmap disallow=all allow=ulaw >> outbound_auth=10001 aors=10001 [10001] type=identify endpoint=10001 >> match=192.168.1.1 when I call 600 from other pbx I getting an notice >> >> NOTICE[10202]: res_pjsip/pjsip_distributor.c:246 >> log_unidentified_request: Request from '"Ilya"' >> failed for '192.168.1.1:5060' (callid: >> ZTNhYjU4ZjU5ZmUxNjM5M2FlYjBlYTE3YzgwZTU4MGY.) - No matching endpoint >> found and "Not Accessable" on phone >> >> let's imagine that 600 its external number of voip operator, and I >> wanna accept all incoming calls from it (no matter what caller id it >> has) what I doing wrong? > > When receiving calls from a VoIP provider you have to match using the source > IP address. You also don't authenticate as the provider will refuse to do so. > > When you control both ends it's really up to you whether to do the matching > based on the source IP address OR use a user account with authentication. If > using the user account the user portion of the From header has to be set to > the username (from_user in pjsip, fromuser in chan_sip). > > Cheers, > > -- > 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 -- > 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] PJSIP outbound register and inbound calls
Hi all, In my case I using realtime, here is how it looks in plant [10001] type=registration transport=upd_static outbound_auth=10001 server_uri=sip:600@192.168.1.1:5060 client_uri=sip:600@192.168.1.4:5060 [10001] type=auth auth_type=userpass password=600 username=600 [10001] type=aor contact=sip:192.168.1.4:5060 [10001] type=endpoint transport=upd_static context=dialmap disallow=all allow=ulaw outbound_auth=10001 aors=10001 [10001] type=identify endpoint=10001 match=192.168.1.1 when I call 600 from other pbx I getting an notice NOTICE[10202]: res_pjsip/pjsip_distributor.c:246 log_unidentified_request: Request from '"Ilya" ' failed for '192.168.1.1:5060' (callid: ZTNhYjU4ZjU5ZmUxNjM5M2FlYjBlYTE3YzgwZTU4MGY.) - No matching endpoint found and "Not Accessable" on phone let's imagine that 600 its external number of voip operator, and I wanna accept all incoming calls from it (no matter what caller id it has) what I 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
