Re: [asterisk-users] Asterisk problems
- Mensagem original - > De: "Jerome SCHEVINGT" > Para: "Asterisk Users Mailing List - Non-Commercial Discussion" > > Enviadas: Quinta-feira, 20 de novembro de 2014 6:18:31 > Assunto: [asterisk-users] Asterisk problems > > > > Hi > > I have a problem with Asterisk 11.5.1. > > When I pick up an incoming phone call sometimes I need to transfer to > someone else in the organization. > I then dial a number on my phone, and press Xfer. > Sometimes it works well, I mean, the number I dialed get the call and > can chat with correspondent. > Sometimes, the number I dialed get the communication and while he chats > with correspondent, gets a bip every 5 sec(incoming call notification). > Just like if I transferred the call twice. > I happens with some internal extensions, not all, but never for others. > Does anyone have already seen this kind of behavior? Please, can you send us a copy-paste of asterisk cli, or logs, showing a sucess transfer and a wrong one? I suspect is a dialplan configuration. Best regards. Alonso. > > > Thanks > Jerome > > > > > > -- > _ > -- 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 problems
Hi I have a problem with Asterisk 11.5.1. When I pick up an incoming phone call sometimes I need to transfer to someone else in the organization. I then dial a number on my phone, and press Xfer. Sometimes it works well, I mean, the number I dialed get the call and can chat with correspondent. Sometimes, the number I dialed get the communication and while he chats with correspondent, gets a bip every 5 sec(incoming call notification). Just like if I transferred the call twice. I happens with some internal extensions, not all, but never for others. Does anyone have already seen this kind of behavior? Thanks Jerome -- _ -- 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 Problems with FXO Ground Start Trunks and DID Wink Start Trunks
I have the following configuration: Stable Asterisk running on a Dell PowerEdge 800 with Enterprise 3 Redhat: Digium TE110P card, connected to a Adtran TA 750 Telco IF: 4 analog DID loop start wink lines, connected to the Adtran FXS card in DPO mode 4 combo analog ground start trunks, connected to the Adtran FXO card in Ground Start Mode. The telco lines and the Adtran channel bank are working. The Digium TE110P card seems to be working also. I can see the bits on the zttool based on changes on the telco lines. All the bits from the Adtran all are correct. However Asterisk does not seem to be setting the correct bits from the software. Problems: 1. Asterisk is not recognizing the incoming DID calls. The CAS bits showing on zttool are correct for the incoming calls, however, Asterisk does not come back with a wink acknowledge. 2. We can receive calls on the FXO ground start channels. However, outbound calls are not working. From the Zttool the idle bits are set fine. However, Asterisk is not setting the correct CAS bits for ground start signaling on a outgoing call on the FXO channel. Has anyone experienced this problem? I have zaptel.conf configured for: span=1,0,0,esf,b8zs fxsgs=1-4 e&m=5-8 loadzone=us defaultzone=us Zapata.conf is configured as: [trunkgroups] [channels] context=default switchtype=national wink=300 rxwink=300 usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 callgroup=1 pickupgroup=1 immediate=no group=1 signaling=fx_gs context=external channel=>1-4 group=2 signaling=em_w context=directindial channel=>5-8 Syed Akbar Alico Systems Inc www.alicosystems.com Tel: 562-436-1510 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk problems behind firewall
1 – 2000 UDP is wrong try 1-2UDP and try port forwarding
rather than opening ports . It probably has to do with the ip of the
server.
mohammed
Jeff R Glassman wrote:
I had my [EMAIL PROTECTED] server working fine SPA-8841 SPA-2100. It was
on an open IP no fire wall. I moved the server behind the firewall.
Now the phones will not dial out. The phones can be called from a DID
or calling to the main POTS number and dialing the extension. However
neither side can hear each other
I opened the following ports in my router
22 TCP
5060 UDP*
5080 UDP*
1 – 2000 UDP
80 TCP
* I have each SPA on different Sip port since they are on the same
network.
Any ideas what else I should do to get it working?
SIP.CONF
[general]
port = 5060 ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)
disallow=all
allow=ulaw
allow=alaw
context = from-sip-external ; Send unknown SIP callers to this context
callerid = Unknown
#include sip_nat.conf
#include sip_additional.conf
sip_additional.conf
205]
username=205
type=friend
secret=[PASSWORD]
qualify=no
port=5060
nat=yes
mailbox=205
host=dynamic
dtmfmode=rfc2833
context=from-internal
canreinvite=no
callerid="Spa-2100 L1" <205
[210]
username=210
type=friend
secret={PASSWORD}
qualify=no
port=5080
nat=yes
mailbox=210
host=dynamic
dtmfmode=rfc2833
context=from-internal
canreinvite=no
callerid="SPA-841" <210
Jeff
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[Asterisk-Users] Asterisk problems behind firewall
I had my [EMAIL PROTECTED] server working fine SPA-8841
SPA-2100. It was on an open IP no fire wall. I moved the server
behind the firewall. Now the phones will not dial out. The phones
can be called from a DID or calling to the main POTS number and dialing the extension.
However neither side can hear each other
I opened the following ports in my router
22 TCP
5060 UDP*
5080 UDP*
1 – 2000 UDP
80
TCP
I have each SPA on different Sip port since they are on
the same network.
Any ideas what else I should do to get it working?
SIP.CONF
[general]
port =
5060 ; Port to bind
to (SIP is 5060)
bindaddr = 0.0.0.0 ; Address to bind to
(all addresses on machine)
disallow=all
allow=ulaw
allow=alaw
context = from-sip-external ; Send unknown SIP callers to
this context
callerid = Unknown
#include sip_nat.conf
#include sip_additional.conf
sip_additional.conf
205]
username=205
type=friend
secret=[PASSWORD]
qualify=no
port=5060
nat=yes
mailbox=205
host=dynamic
dtmfmode=rfc2833
context=from-internal
canreinvite=no
callerid="Spa-2100 L1" <205
[210]
username=210
type=friend
secret={PASSWORD}
qualify=no
port=5080
nat=yes
mailbox=210
host=dynamic
dtmfmode=rfc2833
context=from-internal
canreinvite=no
callerid="SPA-841" <210
Jeff
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[Asterisk-Users] asterisk: problems with connecting to a (german) sip provider
hello ! My problem is: Astriks should create a connection to other members using a german Sip provider (www.sipgate.de). there are no problems with connections to: o Sip- Accounts o national phone numbers o mobile phone numbers but connections to international phone numbers DO NOT WORK (see the attached protokoll). The connection to international phone numbers does work when I directly use a VOIP hardware phone (Grandstream or SNOM). Where is the problem? Does the protokoll give any hint where the problem may be? Thanks for your help for m e an sorry for my bad english. Torsten We're at 80.137.124.154 port 19256 Answering with preferred capability 4 Answering with non-codec capability 1 12 headers, 10 lines Reliably Transmitting: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 80.137.124.154:5060;branch=z9hG4bK710a7bff From: "40" ;tag=as411f3c19 To: Contact: Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Tue, 29 Jun 2004 10:28:37 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 218 v=0 o=root 2645 2645 IN IP4 80.137.124.154 s=session c=IN IP4 80.137.124.154 t=0 0 m=audio 1925878876 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - (NAT) to 217.10.79.9:5060 -- Called [EMAIL PROTECTED] Sip read: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 80.137.124.154:5060;branch=z9hG4bK710a7bff From: "40" ;tag=as411f3c19 To: ;tag=b11cb9bb270104b49a99a995b8c68544.490a Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE Proxy-Authenticate: Digest realm="80.137.124.154", nonce="40e155779906df4b3d4287029d47ac877a53dee9b9fb6" Server: Sip EXpress router (0.8.12-tcp_nonb (i386/linux)) Content-Length: 0 Warning: 392 217.10.79.9:5060 "Noisy feedback tells: pid=7164 req_src_ip=80.137.124.154 req_src_port=5060 in_uri=sip:[EMAIL PROTECTED] out_uri=sip:[EMAIL PROTECTED] via_cnt==1" 10 headers, 0 lines Transmitting: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 80.137.124.154:5060;branch=z9hG4bK710a7bff From: "40" ;tag=as411f3c19 To: ;tag=b11cb9bb270104b49a99a995b8c68544.490a Contact: Call-ID: [EMAIL PROTECTED] CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.10.79.9:5060 We're at 80.137.124.154 port 19256 Answering with preferred capability 4 Answering with non-codec capability 1 Reliably Transmitting: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 80.137.124.154:5060;branch=z9hG4bK710a7bff From: "40" ;tag=as411f3c19 To: Contact: Call-ID: [EMAIL PROTECTED] CSeq: 103 INVITE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="4566565", realm="80.137.124.154", algorithm="MD5", uri="sip:[EMAIL PROTECTED]", nonce="40e1506df4b3d4287029d47ac877a53dee9b9fb6", response="2abfedbcf7681994a0e40ff93fec8534", opaque="" Date: Tue, 29 Jun 2004 10:28:37 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 218 v=0 o=root 2645 2646 IN IP4 80.137.124.154 s=session c=IN IP4 80.137.124.154 t=0 0 m=audio 19256 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - (NAT) to 217.10.79.9:5060 Sip read: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 80.137.124.154:5060;branch=z9hG4bK710a7bff From: "40" ;tag=as411f3c19 To: ;tag=b11cb9bb270104b49a99a995b8c68544.490a Call-ID: [EMAIL PROTECTED] CSeq: 103 INVITE Proxy-Authenticate: Digest realm="80.137.124.154", nonce="40e1506df4b3d4287029d47ac877a53dee9b9fb6" Server: Sip EXpress router (0.8.12-tcp_nonb (i386/linux)) Content-Length: 0 Warning: 392 217.10.79.9:5060 "Noisy feedback tells: pid=7165 req_src_ip=80.137.124.154 req_src_port=5060 in_uri=sip:[EMAIL PROTECTED] out_uri=sip:[EMAIL PROTECTED] via_cnt==1" 10 headers, 0 lines Transmitting: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 80.137.124.154:5060;branch=z9hG4bK710a7bff From: "40" ;tag=as411f3c19 To: ;tag=b11cb9bb270104b49a99a995b8c68544.490a Contact: Call-ID: [EMAIL PROTECTED] CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 217.10.79.9:5060 We're at 80.137.124.154 port 19256 Answering with preferred capability 4 Answering with non-codec capability 1 Reliably Transmitting: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 80.137.124.154:5060;branch=z9hG4bK710a7bff From: "40" ;tag=as411f3c19 To: Contact: Call-ID: [EMAIL PROTECTED] CSeq: 104 INVITE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="4566565", realm="80.137.124.154", algorithm="MD5", uri="sip:[EMAIL PROTECTED]", nonce="40e1506df4b3d4287029d47ac877a53dee9b9fb6", response="2abfedbcf7681994a0e40ff93fec8534", opaque="" Date: Tue, 29 Jun 2004 10:28:37 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 218 v=0 o=root 2645 2647 IN IP4
