PJSIP not sending Authorization Header on 401
I’ve run into an odd issue trying to get my client that is using PJSIP to talk to Asterisk. The first time it tries to REGISTER with my Asterisk server it get’s back a 401 Unauthorized. Which is expected. However the next attempt, PJSIP should send the Authorization Header in it’s REGISTER request. But it’s not doing this. See below for a snippet of a packet capture I’ve done.
10.200.0.72 is my Asterisk server. And 10.200.154.118 is my client. For reference, it’s an iOS project I’m working on and I’m trying to get my device to register as extension 100.
Here’s the really confusing part. This wasn’t working 2 days ago, and then suddenly yesterday it started working fine. Now today, it’s not working again. It’s almost as if it all of sudden starts working and PJSIP finally sends the Authorization Header. Other devices such as my LinkSys PCAP device are able to register without issue. So I know it must be something I’m doing wrong with PJSIP? I’ve been wrestling with this for over a week and I know it must be something simple but it escapes me. Welcome any thoughts anyone might have?
Also, apologies for the length of this message. I wanted to include as much info as possible. I tried to trim down as much as possible.
REGISTER sip:10.200.0.72 SIP/2.0
Via: SIP/2.0/UDP 10.200.154.118:5080;rport;branch=z9hG4bKPjKy2xO6UVBMvX6oiWCt9UrWdclNlhvh31
Max-Forwards: 70
From: sip:100@10.200.0.72;tag=eXXDvPSkJ5WhQUNxs7UNXtHO-rdIO96v
To: sip:100@10.200.0.72
Call-ID: 6ddWjvyiBuGGfEW7vq2AvF-PF8SSpUxZ
CSeq: 13892 REGISTER
Contact: sip:100@10.200.154.118:5080;ob
Expires: 300
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Content-Length: 0
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.200.154.118:5080;branch=z9hG4bKPjKy2xO6UVBMvX6oiWCt9UrWdclNlhvh31;received=174.0.50.116;rport=15923
From: sip:100@10.200.0.72;tag=eXXDvPSkJ5WhQUNxs7UNXtHO-rdIO96v
To: sip:100@10.200.0.72;tag=as45017471
Call-ID: 6ddWjvyiBuGGfEW7vq2AvF-PF8SSpUxZ
CSeq: 13892 REGISTER
Server: Mozilla
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5d4f67d0"
Content-Length: 0
REGISTER sip:10.200.0.72 SIP/2.0
Via: SIP/2.0/UDP 10.200.154.118:5080;rport;branch=z9hG4bKPjKy2xO6UVBMvX6oiWCt9UrWdclNlhvh31
Max-Forwards: 70
From: sip:100@10.200.0.72;tag=eXXDvPSkJ5WhQUNxs7UNXtHO-rdIO96v
To: sip:100@10.200.0.72
Call-ID: 6ddWjvyiBuGGfEW7vq2AvF-PF8SSpUxZ
CSeq: 13892 REGISTER
Contact: sip:100@10.200.154.118:5080;ob
Expires: 300
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Content-Length: 0
REGISTER sip:10.200.0.72 SIP/2.0
Via: SIP/2.0/UDP 10.200.154.118:5080;rport;branch=z9hG4bKPjKy2xO6UVBMvX6oiWCt9UrWdclNlhvh31
Max-Forwards: 70
From: sip:100@10.200.0.72;tag=eXXDvPSkJ5WhQUNxs7UNXtHO-rdIO96v
To: sip:100@10.200.0.72
Call-ID: 6ddWjvyiBuGGfEW7vq2AvF-PF8SSpUxZ
CSeq: 13892 REGISTER
Contact: sip:100@10.200.154.118:5080;ob
Expires: 300
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Content-Length: 0
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.200.154.118:5080;branch=z9hG4bKPjKy2xO6UVBMvX6oiWCt9UrWdclNlhvh31;received=174.0.50.116;rport=15923
From: sip:100@10.200.0.72;tag=eXXDvPSkJ5WhQUNxs7UNXtHO-rdIO96v
To: sip:100@10.200.0.72;tag=as45017471
Call-ID: 6ddWjvyiBuGGfEW7vq2AvF-PF8SSpUxZ
CSeq: 13892 REGISTER
Server: Mozilla
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5d4f67d0"
Content-Length: 0
It’s not any firewall issue obviously as the packets are getting through and I don’t have any restrictions on which IPs can connect to that extension…
For those who are familiar with ObjC.. Here’s my start and register function…
-
(void) startAndRegisterOnServer:(NSString *)sipDomain withUserName:(NSString *)sipUser andPassword:(NSString *)password success:(void (^)(void))success failure:(void (^)(int errorCode, NSString *errorMessage))failure
{
pj_status_t status;status = pjsua_create();
if (status != PJ_SUCCESS)
{
failure (status, @"Error in pjsua_create");return;}
// Setup Config and Initialize
pjsua_config config;
pjsua_config_default (&config);config.cb.on_incoming_call = &on_incoming_call;
config.cb.on_call_media_state = &on_call_media_state;
config.cb.on_call_state = &on_call_state;
config.cb.on_reg_state2 = &on_reg_state2;pjsua_logging_config logConfig;
pjsua_logging_config_default(&logConfig);
logConfig.console_level = 3;status = pjsua_init(&config, &logConfig, NULL);
if (status != PJ_SUCCESS)
{
failure (status, @"Error in pjsua_init");return;}
// Add UDP transport
pjsua_transport_config udpTransportConfig;
pjsua_transport_config_default(&udpTransportConfig);
udpTransportConfig.port = 5080;status = pjsua_transport_create(PJSIP_TRANSPORT_UDP, &udpTransportConfig, NULL);
if (status != PJ_SUCCESS)
{
failure (status, @"Error adding UDP transport");return;}
// Add TCP transport.
pjsua_transport_config tcpTransportConfig;
pjsua_transport_config_default(&tcpTransportConfig);
tcpTransportConfig.port = 5080;status = pjsua_transport_create(PJSIP_TRANSPORT_TCP, &tcpTransportConfig, NULL);
if (status != PJ_SUCCESS)
{
failure (status, @"Error adding TCP transport");return;}
// Startup PJSUA
status = pjsua_start();
if (status != PJ_SUCCESS)
{
failure (status, @"Error starting PJSUA");return;}
// UnRegister with SIP Server First
if (pjsua_acc_get_count() > 0)
{
[self unregisterAccount];
}// Register with SIP Server
pjsua_acc_config accountConfig;
pjsua_acc_config_default(&accountConfig);// Account ID
char sipAccount [MAX_SIP_ACCOUNT_LENGTH];
sprintf (sipAccount, "sip:%s@%s", [sipUser UTF8String], [sipDomain UTF8String]);
accountConfig.id = pj_str(sipAccount);// Register URI
char regUri[MAX_SIP_REGISTER_URI_LENGTH];
sprintf(regUri, "sip:%s", [sipDomain UTF8String]);
accountConfig.reg_uri = pj_str(regUri);// Set the accountConfig with the credentials.
accountConfig.cred_count = 1;
accountConfig.cred_info[0].scheme = pj_str("digest");
accountConfig.cred_info[0].realm = pj_str("asterisk");
accountConfig.cred_info[0].username = pj_str((char *)[sipUser UTF8String]);accountConfig.cred_info[0].data_type = PJSIP_CRED_DATA_PLAIN_PASSWD;
accountConfig.cred_info[0].data = pj_str((char *)[password UTF8String]);
pjsua_acc_id accID;
status = pjsua_acc_add(&accountConfig, PJ_TRUE, &accID);
[PJSIPInterfaceManager sharedInstance].accountID = accID;
if (status != PJ_SUCCESS)
{
failure (status, @"Error registering account with server");return;}
// Shutdown audio for PJSIP since we are using CallKit.
pjsua_set_no_snd_dev();
success();
}
Also I know this isn’t related to PJSIP directly but I’ve included my config settings here on Asterisk for reference…
SIP Settings
sip show settings
Global Settings:
UDP Bindaddress: 0.0.0.0:5060
TCP SIP Bindaddress: 0.0.0.0:5060
TLS SIP Bindaddress: Disabled
Videosupport: No
Textsupport: No
Ignore SDP sess. ver.: No
AutoCreate Peer: Off
Match Auth Username: No
Allow unknown access: Yes
Allow subscriptions: Yes
Allow overlap dialing: Yes
Allow promisc. redir: No
Enable call counters: No
SIP domain support: No
Path support : No
Realm. auth: No
Our auth realm asterisk
Use domains as realms: No
Call to non-local dom.: Yes
URI user is phone no: No
Always auth rejects: Yes
Direct RTP setup: No
User Agent: Mozilla
SDP Session Name: Asterisk PBX 13.7.2
SDP Owner Name: root
Reg. context: (not set)
Regexten on Qualify: No
Trust RPID: No
Send RPID: No
Legacy userfield parse: No
Send Diversion: Yes
Caller ID: Unknown
From: Domain:
Record SIP history: Off
Auth. Failure Events: Off
T.38 support: No
T.38 EC mode: Unknown
T.38 MaxDtgrm: 4294967295
SIP realtime: Disabled
Qualify Freq : 60000 ms
Q.850 Reason header: No
Store SIP_CAUSE: No
Network QoS Settings:
IP ToS SIP: CS3
IP ToS RTP audio: EF
IP ToS RTP video: AF41
IP ToS RTP text: CS0
802.1p CoS SIP: 4
802.1p CoS RTP audio: 5
802.1p CoS RTP video: 6
802.1p CoS RTP text: 5
Jitterbuffer enabled: No
Network Settings:
SIP address remapping: Enabled using externaddr
Externhost: <none>
Externaddr: [REDACTED]
Externrefresh: 10
Localnet: 10.200.0.72/255.255.0.0
Global Signalling Settings:
Codecs: (ulaw|alaw|gsm|g726|g723|g729|speex|g722|g719)
Relax DTMF: No
RFC2833 Compensation: No
Symmetric RTP: Yes
Compact SIP headers: No
RTP Keepalive: 0 (Disabled)
RTP Timeout: 30
RTP Hold Timeout: 300
MWI NOTIFY mime type: application/simple-message-summary
DNS SRV lookup: No
Pedantic SIP support: Yes
Reg. min duration 60 secs
Reg. max duration: 3600 secs
Reg. default duration: 3600 secs
Sub. min duration 60 secs
Sub. max duration: 3600 secs
Outbound reg. timeout: 120 secs
Outbound reg. attempts: 0
Outbound reg. retry 403:0
Notify ringing state: Yes
Include CID: No
Notify hold state: Yes
SIP Transfer mode: open
Max Call Bitrate: 384 kbps
Auto-Framing: No
Outb. proxy: <not set>
Session Timers: Accept
Session Refresher: uas
Session Expires: 1800 secs
Session Min-SE: 90 secs
Timer T1: 500
Timer T1 minimum: 100
Timer B: 32000
No premature media: Yes
Max forwards: 70
Default Settings:
Allowed transports: UDP
Outbound transport: UDP
Context: from-sip-external
Record on feature: automon
Record off feature: automon
Force rport: Yes
DTMF: rfc2833
Qualify: 0
Keepalive: 0
Use ClientCode: No
Progress inband: No
Language:
Tone zone: <Not set>
MOH Interpret: default
MOH Suggest:
Voice Mail Extension: *97
Here is the config for sip peer 100…
sip show peer 100
- Name : 100
Description :
Secret : <Set>
MD5Secret : <Not set>
Remote Secret: <Not set>
Context : from-internal
Record On feature : automon
Record Off feature : automon
Subscr.Cont. : <Not set>
Language :
Tonezone : <Not set>
AMA flags : Unknown
Transfer mode: open
CallingPres : Presentation Allowed, Not Screened
Callgroup :
Pickupgroup :
Named Callgr :
Nam. Pickupgr:
MOH Suggest :
Mailbox : 100@default
VM Extension : *97
LastMsgsSent : 32767/65535
Call limit : 2147483647
Max forwards : 0
Dynamic : Yes
Callerid : "Pete" <100>
MaxCallBR : 384 kbps
Expire : -1
Insecure : no
Force rport : Yes
Symmetric RTP: Yes
ACL : Yes
DirectMedACL : No
T.38 support : No
T.38 EC mode : Unknown
T.38 MaxDtgrm: 4294967295
DirectMedia : No
PromiscRedir : No
User=Phone : No
Video Support: No
Text Support : No
Ign SDP ver : No
Trust RPID : Yes
Send RPID : Yes
Path support : No
Path : N/A
TrustIDOutbnd: Legacy
Subscriptions: Yes
Overlap dial : Yes
DTMFmode : rfc2833
Timer T1 : 500
Timer B : 32000
ToHost :
Addr->IP : (null)
Defaddr->IP : (null)
Prim.Transp. : TCP
Allowed.Trsp : UDP,TCP
Def. Username: 100
SIP Options : (none)
Codecs : (ulaw|alaw|gsm|g726|g723|g729|speex|g722|g719)
Auto-Framing : No
Status : UNKNOWN
Useragent :
Reg. Contact : sip:100@[REDACTED]:54641;ob
Qualify Freq : 60000 ms
Keepalive : 0 ms
Sess-Timers : Accept
Sess-Refresh : uas
Sess-Expires : 1800 secs
Min-Sess : 90 secs
RTP Engine : asterisk
Parkinglot :
Use Reason : No
Encryption : No
I’ve REDACTED any external IP’s for security reasons.
Again, appreciate any thoughts or insights anyone might have. Much appreciated.
Regards,
Peter
I may have found what might be causing my issue. However I’m not sure how to proceed… It appears that Digium (owners of Asterisk) had to patch PJSIP to make things work for them. Have a look at this article…
http://lists.digium.com/pipermail/asterisk-code-review/2015-April/001102.html
This is exactly what I’ve noticed in my packet capture. If you see my original message you’ll note that the CSeq is not incrementing for each request. So Asterisk is thinking its just a retransmittal of the original message.
However, this still does not explain why PJSIP is not sending the Authorization header. Could it be that it is not progressing to the next step and just retransmitting it’s original message for some reason?
Regards,
Peter Warrick
“Don’t Call Us, We’ll Call You."
On Jan 26, 2017, at 5:58 PM, Peter Warrick peter@cabanawire.com wrote:
I’ve run into an odd issue trying to get my client that is using PJSIP to talk to Asterisk. The first time it tries to REGISTER with my Asterisk server it get’s back a 401 Unauthorized. Which is expected. However the next attempt, PJSIP should send the Authorization Header in it’s REGISTER request. But it’s not doing this. See below for a snippet of a packet capture I’ve done.
10.200.0.72 is my Asterisk server. And 10.200.154.118 is my client. For reference, it’s an iOS project I’m working on and I’m trying to get my device to register as extension 100.
Here’s the really confusing part. This wasn’t working 2 days ago, and then suddenly yesterday it started working fine. Now today, it’s not working again. It’s almost as if it all of sudden starts working and PJSIP finally sends the Authorization Header. Other devices such as my LinkSys PCAP device are able to register without issue. So I know it must be something I’m doing wrong with PJSIP? I’ve been wrestling with this for over a week and I know it must be something simple but it escapes me. Welcome any thoughts anyone might have?
Also, apologies for the length of this message. I wanted to include as much info as possible. I tried to trim down as much as possible.
REGISTER sip:10.200.0.72 sip:10.200.0.72 SIP/2.0
Via: SIP/2.0/UDP 10.200.154.118:5080;rport;branch=z9hG4bKPjKy2xO6UVBMvX6oiWCt9UrWdclNlhvh31
Max-Forwards: 70
From: <sip:100@10.200.0.72 sip:100@10.200.0.72>;tag=eXXDvPSkJ5WhQUNxs7UNXtHO-rdIO96v
To: <sip:100@10.200.0.72 sip:100@10.200.0.72>
Call-ID: 6ddWjvyiBuGGfEW7vq2AvF-PF8SSpUxZ
CSeq: 13892 REGISTER
Contact: <sip:100@10.200.154.118:5080;ob sip:100@10.200.154.118:5080;ob>
Expires: 300
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Content-Length: 0
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.200.154.118:5080;branch=z9hG4bKPjKy2xO6UVBMvX6oiWCt9UrWdclNlhvh31;received=174.0.50.116;rport=15923
From: <sip:100@10.200.0.72 sip:100@10.200.0.72>;tag=eXXDvPSkJ5WhQUNxs7UNXtHO-rdIO96v
To: <sip:100@10.200.0.72 sip:100@10.200.0.72>;tag=as45017471
Call-ID: 6ddWjvyiBuGGfEW7vq2AvF-PF8SSpUxZ
CSeq: 13892 REGISTER
Server: Mozilla
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5d4f67d0"
Content-Length: 0
REGISTER sip:10.200.0.72 sip:10.200.0.72 SIP/2.0
Via: SIP/2.0/UDP 10.200.154.118:5080;rport;branch=z9hG4bKPjKy2xO6UVBMvX6oiWCt9UrWdclNlhvh31
Max-Forwards: 70
From: <sip:100@10.200.0.72 sip:100@10.200.0.72>;tag=eXXDvPSkJ5WhQUNxs7UNXtHO-rdIO96v
To: <sip:100@10.200.0.72 sip:100@10.200.0.72>
Call-ID: 6ddWjvyiBuGGfEW7vq2AvF-PF8SSpUxZ
CSeq: 13892 REGISTER
Contact: <sip:100@10.200.154.118:5080;ob sip:100@10.200.154.118:5080;ob>
Expires: 300
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Content-Length: 0
REGISTER sip:10.200.0.72 sip:10.200.0.72 SIP/2.0
Via: SIP/2.0/UDP 10.200.154.118:5080;rport;branch=z9hG4bKPjKy2xO6UVBMvX6oiWCt9UrWdclNlhvh31
Max-Forwards: 70
From: <sip:100@10.200.0.72 sip:100@10.200.0.72>;tag=eXXDvPSkJ5WhQUNxs7UNXtHO-rdIO96v
To: <sip:100@10.200.0.72 sip:100@10.200.0.72>
Call-ID: 6ddWjvyiBuGGfEW7vq2AvF-PF8SSpUxZ
CSeq: 13892 REGISTER
Contact: <sip:100@10.200.154.118:5080;ob sip:100@10.200.154.118:5080;ob>
Expires: 300
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Content-Length: 0
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.200.154.118:5080;branch=z9hG4bKPjKy2xO6UVBMvX6oiWCt9UrWdclNlhvh31;received=174.0.50.116;rport=15923
From: <sip:100@10.200.0.72 sip:100@10.200.0.72>;tag=eXXDvPSkJ5WhQUNxs7UNXtHO-rdIO96v
To: <sip:100@10.200.0.72 sip:100@10.200.0.72>;tag=as45017471
Call-ID: 6ddWjvyiBuGGfEW7vq2AvF-PF8SSpUxZ
CSeq: 13892 REGISTER
Server: Mozilla
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5d4f67d0"
Content-Length: 0
It’s not any firewall issue obviously as the packets are getting through and I don’t have any restrictions on which IPs can connect to that extension…
For those who are familiar with ObjC.. Here’s my start and register function…
-
(void) startAndRegisterOnServer:(NSString *)sipDomain withUserName:(NSString *)sipUser andPassword:(NSString *)password success:(void (^)(void))success failure:(void (^)(int errorCode, NSString *errorMessage))failure
{
pj_status_t status;status = pjsua_create();
if (status != PJ_SUCCESS)
{
failure (status, @"Error in pjsua_create");return;}
// Setup Config and Initialize
pjsua_config config;
pjsua_config_default (&config);config.cb.on_incoming_call = &on_incoming_call;
config.cb.on_call_media_state = &on_call_media_state;
config.cb.on_call_state = &on_call_state;
config.cb.on_reg_state2 = &on_reg_state2;pjsua_logging_config logConfig;
pjsua_logging_config_default(&logConfig);
logConfig.console_level = 3;status = pjsua_init(&config, &logConfig, NULL);
if (status != PJ_SUCCESS)
{
failure (status, @"Error in pjsua_init");return;}
// Add UDP transport
pjsua_transport_config udpTransportConfig;
pjsua_transport_config_default(&udpTransportConfig);
udpTransportConfig.port = 5080;status = pjsua_transport_create(PJSIP_TRANSPORT_UDP, &udpTransportConfig, NULL);
if (status != PJ_SUCCESS)
{
failure (status, @"Error adding UDP transport");return;}
// Add TCP transport.
pjsua_transport_config tcpTransportConfig;
pjsua_transport_config_default(&tcpTransportConfig);
tcpTransportConfig.port = 5080;status = pjsua_transport_create(PJSIP_TRANSPORT_TCP, &tcpTransportConfig, NULL);
if (status != PJ_SUCCESS)
{
failure (status, @"Error adding TCP transport");return;}
// Startup PJSUA
status = pjsua_start();
if (status != PJ_SUCCESS)
{
failure (status, @"Error starting PJSUA");return;}
// UnRegister with SIP Server First
if (pjsua_acc_get_count() > 0)
{
[self unregisterAccount];
}// Register with SIP Server
pjsua_acc_config accountConfig;
pjsua_acc_config_default(&accountConfig);// Account ID
char sipAccount [MAX_SIP_ACCOUNT_LENGTH];
sprintf (sipAccount, "sip:%s@%s sip:%s@%s", [sipUser UTF8String], [sipDomain UTF8String]);
accountConfig.id = pj_str(sipAccount);// Register URI
char regUri[MAX_SIP_REGISTER_URI_LENGTH];
sprintf(regUri, "sip:%s sip:%s", [sipDomain UTF8String]);
accountConfig.reg_uri = pj_str(regUri);// Set the accountConfig with the credentials.
accountConfig.cred_count = 1;
accountConfig.cred_info[0].scheme = pj_str("digest");
accountConfig.cred_info[0].realm = pj_str("asterisk");
accountConfig.cred_info[0].username = pj_str((char *)[sipUser UTF8String]);accountConfig.cred_info[0].data_type = PJSIP_CRED_DATA_PLAIN_PASSWD;
accountConfig.cred_info[0].data = pj_str((char *)[password UTF8String]);
pjsua_acc_id accID;
status = pjsua_acc_add(&accountConfig, PJ_TRUE, &accID);
[PJSIPInterfaceManager sharedInstance].accountID = accID;
if (status != PJ_SUCCESS)
{
failure (status, @"Error registering account with server");return;}
// Shutdown audio for PJSIP since we are using CallKit.
pjsua_set_no_snd_dev();
success();
}
Also I know this isn’t related to PJSIP directly but I’ve included my config settings here on Asterisk for reference…
SIP Settings
sip show settings
Global Settings:
UDP Bindaddress: 0.0.0.0:5060
TCP SIP Bindaddress: 0.0.0.0:5060
TLS SIP Bindaddress: Disabled
Videosupport: No
Textsupport: No
Ignore SDP sess. ver.: No
AutoCreate Peer: Off
Match Auth Username: No
Allow unknown access: Yes
Allow subscriptions: Yes
Allow overlap dialing: Yes
Allow promisc. redir: No
Enable call counters: No
SIP domain support: No
Path support : No
Realm. auth: No
Our auth realm asterisk
Use domains as realms: No
Call to non-local dom.: Yes
URI user is phone no: No
Always auth rejects: Yes
Direct RTP setup: No
User Agent: Mozilla
SDP Session Name: Asterisk PBX 13.7.2
SDP Owner Name: root
Reg. context: (not set)
Regexten on Qualify: No
Trust RPID: No
Send RPID: No
Legacy userfield parse: No
Send Diversion: Yes
Caller ID: Unknown
From: Domain:
Record SIP history: Off
Auth. Failure Events: Off
T.38 support: No
T.38 EC mode: Unknown
T.38 MaxDtgrm: 4294967295
SIP realtime: Disabled
Qualify Freq : 60000 ms
Q.850 Reason header: No
Store SIP_CAUSE: No
Network QoS Settings:
IP ToS SIP: CS3
IP ToS RTP audio: EF
IP ToS RTP video: AF41
IP ToS RTP text: CS0
802.1p CoS SIP: 4
802.1p CoS RTP audio: 5
802.1p CoS RTP video: 6
802.1p CoS RTP text: 5
Jitterbuffer enabled: No
Network Settings:
SIP address remapping: Enabled using externaddr
Externhost: <none>
Externaddr: [REDACTED]
Externrefresh: 10
Localnet: 10.200.0.72/255.255.0.0
Global Signalling Settings:
Codecs: (ulaw|alaw|gsm|g726|g723|g729|speex|g722|g719)
Relax DTMF: No
RFC2833 Compensation: No
Symmetric RTP: Yes
Compact SIP headers: No
RTP Keepalive: 0 (Disabled)
RTP Timeout: 30
RTP Hold Timeout: 300
MWI NOTIFY mime type: application/simple-message-summary
DNS SRV lookup: No
Pedantic SIP support: Yes
Reg. min duration 60 secs
Reg. max duration: 3600 secs
Reg. default duration: 3600 secs
Sub. min duration 60 secs
Sub. max duration: 3600 secs
Outbound reg. timeout: 120 secs
Outbound reg. attempts: 0
Outbound reg. retry 403:0
Notify ringing state: Yes
Include CID: No
Notify hold state: Yes
SIP Transfer mode: open
Max Call Bitrate: 384 kbps
Auto-Framing: No
Outb. proxy: <not set>
Session Timers: Accept
Session Refresher: uas
Session Expires: 1800 secs
Session Min-SE: 90 secs
Timer T1: 500
Timer T1 minimum: 100
Timer B: 32000
No premature media: Yes
Max forwards: 70
Default Settings:
Allowed transports: UDP
Outbound transport: UDP
Context: from-sip-external
Record on feature: automon
Record off feature: automon
Force rport: Yes
DTMF: rfc2833
Qualify: 0
Keepalive: 0
Use ClientCode: No
Progress inband: No
Language:
Tone zone: <Not set>
MOH Interpret: default
MOH Suggest:
Voice Mail Extension: *97
Here is the config for sip peer 100…
sip show peer 100
- Name : 100
Description :
Secret : <Set>
MD5Secret : <Not set>
Remote Secret: <Not set>
Context : from-internal
Record On feature : automon
Record Off feature : automon
Subscr.Cont. : <Not set>
Language :
Tonezone : <Not set>
AMA flags : Unknown
Transfer mode: open
CallingPres : Presentation Allowed, Not Screened
Callgroup :
Pickupgroup :
Named Callgr :
Nam. Pickupgr:
MOH Suggest :
Mailbox : 100@default
VM Extension : *97
LastMsgsSent : 32767/65535
Call limit : 2147483647
Max forwards : 0
Dynamic : Yes
Callerid : "Pete" <100>
MaxCallBR : 384 kbps
Expire : -1
Insecure : no
Force rport : Yes
Symmetric RTP: Yes
ACL : Yes
DirectMedACL : No
T.38 support : No
T.38 EC mode : Unknown
T.38 MaxDtgrm: 4294967295
DirectMedia : No
PromiscRedir : No
User=Phone : No
Video Support: No
Text Support : No
Ign SDP ver : No
Trust RPID : Yes
Send RPID : Yes
Path support : No
Path : N/A
TrustIDOutbnd: Legacy
Subscriptions: Yes
Overlap dial : Yes
DTMFmode : rfc2833
Timer T1 : 500
Timer B : 32000
ToHost :
Addr->IP : (null)
Defaddr->IP : (null)
Prim.Transp. : TCP
Allowed.Trsp : UDP,TCP
Def. Username: 100
SIP Options : (none)
Codecs : (ulaw|alaw|gsm|g726|g723|g729|speex|g722|g719)
Auto-Framing : No
Status : UNKNOWN
Useragent :
Reg. Contact : sip:100@[REDACTED]:54641;ob sip:100@[REDACTED]:54641;ob
Qualify Freq : 60000 ms
Keepalive : 0 ms
Sess-Timers : Accept
Sess-Refresh : uas
Sess-Expires : 1800 secs
Min-Sess : 90 secs
RTP Engine : asterisk
Parkinglot :
Use Reason : No
Encryption : No
I’ve REDACTED any external IP’s for security reasons.
Again, appreciate any thoughts or insights anyone might have. Much appreciated.
Regards,
Peter
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