Can't make outgoing call with Skype Connect
I have my Asterisk PBX configured with Skype Connect using SIP with TLS and SRTP. Most of my outgoing calls go through, but sometimes I can't get call out. I was able to leave asterisk console up and collect verbose and sip debug data. Can somebody help me diagnose why my calls aren't going through?
I've changed my external IP (I'm behind a NAT'd firewall) to 1.2.3.4 and my SIP profile's user ID to 11111111111111. and my domain name to test.com. If someone working for Skype needs that information they can email me and I'll send it privately.
My config:
[general]
context=default_context
allowguest=no
alwaysauthreject=yes
allowoverlap=no
udpbindaddr=0.0.0.0
tlsenable=yes
tlsbinddir=0.0.0.0
tlscertfile=/usr/local/asterisk/etc/asterisk/keys/asterisk.pem
tlscafile=/usr/local/asterisk/etc/asterisk/keys/ca.crt
tlscipher=ALL
tlsclientmethod=tlsv1
tcpenable=yes
tcpbindaddr=0.0.0.0
transport=udp,tcp,tls
srvlookup=yes
dynamic_exclude_static = yes
buggymwi=yes
contactpermit=192.168.1.0/24
register => tls://111111111111111:[email protected]
[skype]
type=friend
context=from-skype
dtmfmode=rfc2833
host=sip.skype.com
username=11111111111111
fromuser=11111111111111
secret=abcd12345
disallow=all
allow=ulaw
allow=alaw
nat=yes
fromdomain=sip.skype.com
insecure=port,invite
transport=tls
srtpcapable=yes
encryption=yes
SIP Debugging enabled
[2012-08-23 19:22:33] NOTICE[16552]: chan_sip.c:13465 sip_reregister: -- Re-registration for [email protected]
> doing dnsmgr_lookup for 'sip.skype.com'
> ast_get_srv: SRV lookup for '_sips._tcp.sip.skype.com' mapped to host 1.sip.skype.com, port 5061
REGISTER 11 headers, 0 lines
Reliably Transmitting (NAT) to 63.209.144.201:5061:
REGISTER sip:sip.skype.com:5061 SIP/2.0
Via: SIP/2.0/TLS 192.168.1.15:5061;branch=z9hG4bK2726fcb8;rport
Max-Forwards: 70
From: <sip:[email protected]>;tag=as6edf93cf
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 32495 REGISTER
User-Agent: Asterisk PBX 10.5.2
Authorization: Digest username="11111111111111", realm="sip.skype.com", algorithm=MD5, uri="sip:sip.skype.com:5061", nonce="5036b5770000182c78c1d1909cfd5c74e33f033c952d240d", response="81001ceacd91b16ebb956d3c55991471"
Expires: 120
Contact: <sip:[email protected]:5061;transport=TLS>
Content-Length: 0
<--- SIP read from TLS:63.209.144.201:5061 --->
SIP/2.0 200 OK
From: <sip:[email protected]>;tag=as6edf93cf
To: <sip:[email protected]>;tag=c990d13f-90f7a10d-0-55cb59a8-0
Call-ID: [email protected]
CSeq: 32495 REGISTER
Via: SIP/2.0/TLS 192.168.1.15:5061;branch=z9hG4bK2726fcb8;rport=50541;received=1.2.3.4
Expires: 45
Contact: <sip:[email protected]:5061;transport=tls>;expires=45
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: REGISTER)
[2012-08-23 19:22:33] NOTICE[17932]: chan_sip.c:21399 handle_response_register: Outbound Registration: Expiry for sip.skype.com is 45 sec (Scheduling reregistration in 30 s)
<--- SIP read from UDP:192.168.1.16:5060 --->
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.16:5060;branch=z9hG4bK-819ce62
From: "Scott's Office" <sip:[email protected]>;tag=9961686dacab532ao0
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 101 INVITE
Max-Forwards: 70
Contact: "Scott's Office" <sip:[email protected]:5060>
Expires: 240
User-Agent: Cisco/SPA504G-7.5.2b
Content-Length: 234
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE
Supported: replaces
Content-Type: application/sdp
v=0
o=- 88651316 88651316 IN IP4 192.168.1.16
s=-
c=IN IP4 192.168.1.16
t=0 0
m=audio 16484 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
<------------->
--- (14 headers 12 lines) ---
Sending to 192.168.1.16:5060 (NAT)
Using INVITE request as basis request - [email protected]
Found peer 'scott_office' for 'scott_office' from 192.168.1.16:5060
== Using SIP RTP CoS mark 5
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.1.16:16484
Looking for 19739928881 in home (domain asterisk.test.com)
list_route: hop: <sip:[email protected]:5060>
<--- Transmitting (NAT) to 192.168.1.16:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.16:5060;branch=z9hG4bK-819ce62;received=192.168.1.16;rport=5060
From: "Scott's Office" <sip:[email protected]>;tag=9961686dacab532ao0
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 101 INVITE
Server: Asterisk PBX 10.5.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:[email protected]:5060>
Content-Length: 0
<------------>
-- Executing [19739928881@home:1] Dial("SIP/scott_office-000000b0", "SIP/skype/+19739928881") in new stack
== Using SIP RTP CoS mark 5
Audio is at 9302
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 63.209.144.201:5061:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/TLS 192.168.1.15:5061;branch=z9hG4bK1c3f16ee;rport
Max-Forwards: 70
From: "Scott's Office" <sip:[email protected]:5060>;tag=as049830bd
To: <sip:[email protected]>
Contact: <sip:[email protected]:5061;transport=TLS>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX 10.5.2
Date: Thu, 23 Aug 2012 23:22:34 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 370
v=0
o=root 1671301052 1671301052 IN IP4 192.168.1.15
s=Asterisk PBX 10.5.2
c=IN IP4 192.168.1.15
t=0 0
m=audio 9302 RTP/SAVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:TRh/HeKozlBO/mmYHNTiS5KMnefVI0aRicLoDNjb
-- Called SIP/skype/+19739928881
<--- SIP read from TLS:63.209.144.201:5061 --->
SIP/2.0 100 Trying
From: "Scott's Office" <sip:[email protected]:5060>;tag=as049830bd
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 INVITE
Via: SIP/2.0/TLS 192.168.1.15:5061;branch=z9hG4bK1c3f16ee;rport=50541;received=1.2.3.4
Contact: <sip:[email protected]:5061;maddr=63.209.144.201;transport=tls>
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
<--- SIP read from TLS:63.209.144.201:5061 --->
SIP/2.0 407 Proxy Authentication Required
From: "Scott's Office" <sip:[email protected]:5060>;tag=as049830bd
To: <sip:[email protected]>;tag=c990d13f-13c4-5036bb3b-714198b9-32d150b4
Call-ID: [email protected]
CSeq: 102 INVITE
Proxy-Authenticate: Digest realm="sip.skype.com", nonce="5036bb5800012cdd3d20e5090cc200805f7d0bbd58318e9e", algorithm=MD5
Via: SIP/2.0/TLS 192.168.1.15:5061;branch=z9hG4bK1c3f16ee;rport=50541;received=1.2.3.4
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
set_destination: Parsing <sip:[email protected]> for address/port to send to
set_destination: set destination to 63.209.144.201:5060
Transmitting (NAT) to 63.209.144.201:5061:
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/TLS 192.168.1.15:5061;branch=z9hG4bK1c3f16ee;rport
Max-Forwards: 70
From: "Scott's Office" <sip:[email protected]:5060>;tag=as049830bd
To: <sip:[email protected]>;tag=c990d13f-13c4-5036bb3b-714198b9-32d150b4
Contact: <sip:[email protected]:5061;transport=TLS>
Call-ID: [email protected]
CSeq: 102 ACK
User-Agent: Asterisk PBX 10.5.2
Content-Length: 0
Audio is at 9302
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 63.209.144.201:5061:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/TLS 192.168.1.15:5061;branch=z9hG4bK1b988ba5;rport
Max-Forwards: 70
From: "Scott's Office" <sip:[email protected]:5060>;tag=as049830bd
To: <sip:[email protected]>
Contact: <sip:[email protected]:5061;transport=TLS>
Call-ID: [email protected]
CSeq: 103 INVITE
User-Agent: Asterisk PBX 10.5.2
Proxy-Authorization: Digest username="11111111111111", realm="sip.skype.com", algorithm=MD5, uri="sip:[email protected]", nonce="5036bb5800012cdd3d20e5090cc200805f7d0bbd58318e9e", response="6efb0e37178bae868f0a1e0ddf110e3c"
Date: Thu, 23 Aug 2012 23:22:34 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 370
v=0
o=root 1671301052 1671301053 IN IP4 192.168.1.15
s=Asterisk PBX 10.5.2
c=IN IP4 192.168.1.15
t=0 0
m=audio 9302 RTP/SAVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:TRh/HeKozlBO/mmYHNTiS5KMnefVI0aRicLoDNjb
<--- SIP read from TLS:63.209.144.201:5061 --->
SIP/2.0 100 Trying
From: "Scott's Office" <sip:[email protected]:5060>;tag=as049830bd
To: <sip:[email protected]>;tag=c990d13f-13c4-5036bb3b-714198b9-32d150b4
Call-ID: [email protected]
CSeq: 103 INVITE
Via: SIP/2.0/TLS 192.168.1.15:5061;branch=z9hG4bK1b988ba5;rport=50541;received=1.2.3.4
Contact: <sip:[email protected]:5061;maddr=63.209.144.201;transport=tls>
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '[email protected]' Method: REGISTER
<--- SIP read from TLS:63.209.144.201:5061 --->
SIP/2.0 180 Ringing
From: "Scott's Office" <sip:[email protected]:5060>;tag=as049830bd
To: <sip:[email protected]>;tag=c990d13f-13c4-5036bb3b-714198b9-32d150b4
Call-ID: [email protected]
CSeq: 103 INVITE
User-Agent: SipGW 8
Via: SIP/2.0/TLS 192.168.1.15:5061;branch=z9hG4bK1b988ba5;rport=50541;received=1.2.3.4
Contact: <sip:[email protected]:5061;maddr=63.209.144.201;transport=tls>
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
list_route: hop: <sip:[email protected]:5061;maddr=63.209.144.201;transport=tls>
-- SIP/skype-000000b1 is ringing
<--- Transmitting (NAT) to 192.168.1.16:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.16:5060;branch=z9hG4bK-819ce62;received=192.168.1.16;rport=5060
From: "Scott's Office" <sip:[email protected]>;tag=9961686dacab532ao0
To: <sip:[email protected]>;tag=as3f27fa61
Call-ID: [email protected]
CSeq: 101 INVITE
Server: Asterisk PBX 10.5.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:[email protected]:5060>
Content-Length: 0
<------------>
<--- SIP read from TLS:63.209.144.201:5061 --->
SIP/2.0 408 Request Timeout
From: "Scott's Office" <sip:[email protected]:5060>;tag=as049830bd
To: <sip:[email protected]>;tag=c990d13f-13c4-5036bb3b-714198b9-32d150b4
Call-ID: [email protected]
CSeq: 103 INVITE
Via: SIP/2.0/TLS 192.168.1.15:5061;branch=z9hG4bK1b988ba5;rport=50541;received=1.2.3.4
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
[2012-08-23 19:22:45] WARNING[17932]: chan_sip.c:20947 handle_response_invite: Re-invite to non-existing call leg on other UA. SIP dialog '[email protected]'. Giving up.
set_destination: Parsing <sip:[email protected]> for address/port to send to
set_destination: set destination to 63.209.144.201:5060
Transmitting (NAT) to 63.209.144.201:5061:
ACK sip:[email protected]:5061;maddr=63.209.144.201;transport=tls SIP/2.0
Via: SIP/2.0/TLS 192.168.1.15:5061;branch=z9hG4bK1b988ba5;rport
Max-Forwards: 70
From: "Scott's Office" <sip:[email protected]:5060>;tag=as049830bd
To: <sip:[email protected]>;tag=c990d13f-13c4-5036bb3b-714198b9-32d150b4
Contact: <sip:[email protected]:5061;transport=TLS>
Call-ID: [email protected]
CSeq: 103 ACK
User-Agent: Asterisk PBX 10.5.2
Content-Length: 0
Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: INVITE)
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing [19739928881@home:2] Hangup("SIP/scott_office-000000b0", "") in new stack
== Spawn extension (home, 19739928881, 2) exited non-zero on 'SIP/scott_office-000000b0'
Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: INVITE)
<--- Reliably Transmitting (NAT) to 192.168.1.16:5060 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.1.16:5060;branch=z9hG4bK-819ce62;received=192.168.1.16;rport=5060
From: "Scott's Office" <sip:[email protected]>;tag=9961686dacab532ao0
To: <sip:[email protected]>;tag=as3f27fa61
Call-ID: [email protected]
CSeq: 101 INVITE
Server: Asterisk PBX 10.5.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
<------------>
<--- SIP read from UDP:192.168.1.16:5060 --->
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.16:5060;branch=z9hG4bK-819ce62
From: "Scott's Office" <sip:[email protected]>;tag=9961686dacab532ao0
To: <sip:[email protected]>;tag=as3f27fa61
Call-ID: [email protected]
CSeq: 101 ACK
Max-Forwards: 70
Contact: "Scott's Office" <sip:[email protected]:5060>
User-Agent: Cisco/SPA504G-7.5.2b
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '[email protected]' Method: INVITE
Really destroying SIP dialog '[email protected]' Method: ACK
I wound up calling skype support. This is the final sip.conf looks like. Hope it helps. Good luck.
Scott
[general]
context=default_context
allowguest=no
alwaysauthreject=yes
allowoverlap=no
udpbindaddr=0.0.0.0
tlsenable=yes
tlsbinddir=0.0.0.0
tlscertfile=/usr/local/asterisk/etc/asterisk/keys/asterisk.pem
tlscafile=/usr/local/asterisk/etc/asterisk/keys/ca.crt
tlscipher=ALL
tlsclientmethod=tlsv1
tcpenable=yes
tcpbindaddr=0.0.0.0
transport=udp,tcp,tls
srvlookup=yes
dynamic_exclude_static = yes
buggymwi=yes
contactpermit=192.168.1.0/24
register => tls://[email protected]
[skype]
type=friend
context=from-skype
dtmfmode=rfc2833
host=sip.skype.com
username=user
fromuser=user
secret=pass
disallow=all
allow=ulaw
allow=alaw
nat=yes
fromdomain=sip.skype.com
insecure=port,invite
transport=tls
srtpcapable=yes
encryption=yes
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I live in Burundi, with the country code +257. I can make calls to the US, Canada and Europe but I can't make local calls. I have tried all the troubleshooting tips and tricks, checked my settings etc. Any ideas?
You also posted this on an iPhone forum. Not sure why you posted it here on an iPad forum.
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Can't make phone calls with Siri
I can't make phone call through siri. I'm from Myanmar and local numbers can't be called. Whenever I call, siri said "sorry, I can't make phone calls using that number". I added my country code +95 but it didn't help. But when I call oversea numbers, it works just fine.
Handoff System preferences.
Handoff Continuity Troubleshooting -
Problem - Can't make phone calls with my new Tour 9630
I just bought a new phone - Tour 9630. Its a brand new phone on Sprint.
Now I am in India using this phone on Vodafone and all the services are on, I can send and receive emails, texts, go to websites and download stuff BUT I cant make any calls from that phone.
Everytime I try making a call, it just disconnects after 2 seconds as if it was not registering anything.
What I tried:
Change the mobile connection to GSM only and hard reset the phone - DID NOT WORK
I tried calling service provider who mentioned all my services are working fine
I tried changing the sim card on the new phone - DID NOT WORK
I used my Vodafone sim in another - Works fine but not on the new phone.
I called blackberry support at vodafone - they say I might have to update the OS, which does not sound as music to my ears because this a new phone.
I rebooted the phone million times - DID NOT WORK
I wiped the phone and reinstalled everything - DID NOT WORK
Checked and changed all the setting and tried making calls - DID NOT WORK
Anything you could do to please help me so that I can make phone calls will be highly appreciated.My issue has been taken care of. It seems that Sprint has had an issue with the phone number prefix 238 in the 24151 zipcode. I ended up having to change my phone number and put my phone number under a new zipcode. It all works awesome again. Thanks.
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Can't make a call with my 8130
I just noticed today that I can't make calls. Everytime I try to make a call it says "only emergency calls are allowed"
I can send an SMS and when I try to call my cell from another phone it goes to the msg after one ring but my BB wont ring thoughOnly emergency calls allowed usually indicates a telecom carrier issue. Make sure the indicator near your signal strength is in capitals letters, it should be something like GPRS, EDGE, or 1XEV. If the indicator is in lower case, it means the BB has detected the carrier, but is not being allowed into their network. The telecoms are required by law to allow 911 calls to go through, which accounts for the message you are seeing. If you are roaming, it may be that your home telecom does not have a roaming agreement with the telecom you are currently latched onto.
A couple of things to try:
-Pop the battery out for a few seconds, then put it back in. This will reset the device and all connections, forcing it to reconnect again. This will often fix errors caused by a bad tower connection.
-If you are in your home service area, try resending the registration signal using Options/Advanced Options/Host Routing Table, click for the menu and select Register Now
-If you are roaming, see if there are other carriers in the area you are in using Options/Mobile Network
If none of these work, you are probably going to have to contact your carrier.
posted by DigitalFrog
WARNING: May contain traces of nuts. -
Receiving incoming phone calls but can't make outgoing calls
while connected to my vehicle's hands free device my outgoing call was dropped...when I tryed to redial nothing occurred to connect to the number; once home tried to make a call free from the hands free and same results; however, I'm able to receive calls to the device.
if your on sprint, they are experiencing a nationwide outage with certain ios devices, no ETA available, no work around as of now
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