Call from Linphone to Cisco-VG202 & Cisco 2801

datta.parth · November 22, 2023, 5:37am

Hi,
I have two Cisco Voice router VG202 and Cisco 2801. I have established calls between two Voice router using H323. User can call to each other from both the end.
The two router cisco and svetah is registered to Asteris-21.0.0
Endpoint: 6001 Unavailable 0 of inf
InAuth: 6001/6001
Aor: 6001 1

Endpoint: 6003 Unavailable 0 of inf
InAuth: 6003/6003
Aor: 6003 1

Endpoint: 6004/6004 Not in use 0 of inf
InAuth: 6004/6004
Aor: 6004 3
Contact: 6004/sip:6004@10.128.1.86:42521;transport= a8503c3f96 NonQual nan

Endpoint: cisco Not in use 0 of inf
Aor: cisco 1
Contact: cisco/sip:cisco@10.128.120.5 46ea90a3a9 NonQual nan
Identify: cisco/cisco
Match: 10.128.120.5/32

Endpoint: svetah Not in use 0 of inf
Aor: svetah 1
Contact: svetah/sip:svetah@172.20.1.2 f2b45a28f6 NonQual nan
Identify: svetah/svetah
Match: 172.20.1.2/32
One of the Sip client 6004 is also registered to to Asterisk as shown above using “Linphone”
The Linphone user can call cisco-2801 voice port and vice versa.
User from VG202 can call linphone user on extension 6004 but the linphone user is not able to call Cisco-VG202 user. I have run the debug command on Cisco VG202 and here is what I see:
svetah-voip#
Nov 21 11:10:34.618: //-1/xxxxxxxxxxxx/CCAPI/cc_set_post_tagdata:
CALL_ERROR; Avlist Set Is Failed
Nov 21 11:10:34.626: //-1/xxxxxxxxxxxx/CCAPI/cc_validate_mlpp_info:
Unsupported MLPP Service Domain Network 0
Nov 21 11:10:34.626: //-1/xxxxxxxxxxxx/CCAPI/cc_set_mlpp_info:
Invalid MLPP Information
Nov 21 11:10:34.626: //-1/xxxxxxxxxxxx/CCAPI/cc_validate_mlpp_info:
Unsupported MLPP Service Domain Network 0
Nov 21 11:10:34.626: //-1/xxxxxxxxxxxx/CCAPI/cc_set_mlpp_info:
Invalid MLPP Information
Nov 21 11:10:34.626: //-1/xxxxxxxxxxxx/CCAPI/cc_validate_mlpp_info:
Unsupported MLPP Service Domain Network 0
svetah-voip#
Nov 21 11:10:34.626: //-1/xxxxxxxxxxxx/CCAPI/cc_set_mlpp_info:
Invalid MLPP Information
Nov 21 11:10:34.630: //-1/xxxxxxxxxxxx/CCAPI/cc_set_outpulsed_digits:
Invalid Arguments
Nov 21 11:10:34.638: //-1/xxxxxxxxxxxx/CCAPI/cc_exchange_calls_mlpp_info:
Unable too Get MLPP nformation for callID 137
svetah-voip#
Nov 21 11:10:38.118: //-1/xxxxxxxxxxxx/CCAPI/cc_set_post_tagdata:
CALL_ERROR; Avlist Set Is Failed
Nov 21 11:10:38.122: //-1/xxxxxxxxxxxx/CCAPI/cc_validate_mlpp_info:
Unsupported MLPP Service Domain Network 0
Nov 21 11:10:38.122: //-1/xxxxxxxxxxxx/CCAPI/cc_set_mlpp_info:
Invalid MLPP Information
Nov 21 11:10:38.126: //-1/xxxxxxxxxxxx/CCAPI/cc_validate_mlpp_info:
Unsupported MLPP Service Domain Network 0
Nov 21 11:10:38.126: //-1/xxxxxxxxxxxx/CCAPI/cc_set_mlpp_info:
Invalid MLPP Information
Nov 21 11:10:38.126: //-1/xxxxxxxxxxxx/CCAPI/cc_validate_mlpp_info:
Unsupported MLPP Service Domain Network 0
svetah-voip#
Nov 21 11:10:38.126: //-1/xxxxxxxxxxxx/CCAPI/cc_set_mlpp_info:
Invalid MLPP Information
Nov 21 11:10:38.130: //-1/xxxxxxxxxxxx/CCAPI/cc_set_outpulsed_digits:
Invalid Arguments
Nov 21 11:10:38.138: //-1/xxxxxxxxxxxx/CCAPI/cc_exchange_calls_mlpp_info:
Unable too Get MLPP nformation for callID 139
######################
extension.conf
[from-internal]
exten = 5000,1,Goto(IeplMenu,start,1)

[from-internal]
; Internal calls
exten => _6XXX,1,NoOp(Internal call to extension ${EXTEN}) ; Dial the user extension starting with 6
same => n,Dial(PJSIP/${EXTEN},30) ; Dail and with to answer for 30 second
same => n,Voicemail(${EXTEN}@default,u)
same => n,Hangup() ; Hangup the call
exten => _8.,1,Dial(PJSIP/${EXTEN}@cisco,30) ; Dial the extension in cisco-2801 router
same => n,Hangup()
exten => _7.,1,Dial(PJSIP/${EXTEN}@svetah,30) ; Dial the extension in cisco-2801 router
same => n,Hangup()

[IeplMenu]
exten => start,1,Answer()
same => n,Wait(1) ; Wait for 1 second
same => n,Playback(welcome) ; Play a welcome message
same => n,Wait(1) ; Wait for 1 second
same => n,Background(if-u-know-ext-dial) ; Play a menu of options
same => n,WaitExten(10) ; Wait for the caller to enter an option within 10 seconds
exten => _6XXX,1,NoOp(Internal call to extension ${EXTEN}); Allow the user to dial internal extension
same => n,Dial(PJSIP/${EXTEN},30,) ; Dial the extension, if unanswered go to Playback
same => n,Playback(vm-nobodyavail) ; Play back the message
same => n,Hangup() ; Hangup the call
exten => 8000,1,Dial(PJSIP/8000@cisco,20) ; Allow the user to dial the enxtension 8000
same => n,Playback(vm-nobodyavail) ; If the call is unanswered with in 15 sec go to Voice mail
same => n,Hangup() ; Hangup the call
exten => 7000,1,Dial(PJSIP/7000@svetah,20) ; Allow the user to dial the enxtension 8000
same => n,Playback(vm-nobodyavail) ; If the call is unanswered with in 15 sec go to Voice mail
same => n,Hangup() ; Hangup the call

exten => i,1,Playback(pbx-invalid) ; Play back invaldi extension if the dail plan is not avalable
same => n,Goto(IeplMenu,start,1) ; Go back to TestMenu

;exten => t,1,Playback(please-try-again) ; Try again if invalid extension is pressed.
;same => n,Goto(IeplMenu,start,1) ; Go back to TestMenu

exten => t,1,Playback(vm-goodbye) ; Try again if invalid extension is pressed.
same => n,Handup()

Requesting your help.

penguinpbx · December 12, 2023, 4:40pm

Why not SIP ?

This could be codec mismatch issue. You might try limiting what is allowed to basic ulaw or alaw, on all sides (Asterisk, Linphone and Cisco.)

Also posting the Asterisk logs could perhaps get more feedback from others in these Asterisk-specific forums.

datta.parth · December 13, 2023, 5:29am

Hello,
Thank you for the reply. Here is the Asterisk log: I am using ulaw here is my pjsip.conf
[svetah]
type=endpoint
context=from-internal
disallow=all
allow=ulaw
direct_media=no
aors=svetah
callerid=“svetah”
[svetah]
type=aor
contact=sip:svetah@172.20.1.2
max_contacts=1
[svetah]
type=identify
endpoint=svetah
match=172.20.1.2
####### Log from asterisk####
<— Received SIP request (1310 bytes) from UDP:10.128.1.55:40340 —>
INVITE sip:7000@10.128.110.26 SIP/2.0
Via: SIP/2.0/UDP 10.128.1.55:40340;branch=z9hG4bK.OjyEIFqW8;rport
From: sip:6004@10.128.110.26;tag=mpXmZR2eX
To: sip:7000@10.128.110.26
CSeq: 20 INVITE
Call-ID: cCvlGJ2hTN
Max-Forwards: 70
Supported: replaces, outbound, gruu, path, record-aware
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, PRACK, UPDATE
Content-Type: application/sdp
Content-Length: 684
Contact: sip:6004@10.128.1.55:40340;transport=udp;expires=3599;+org.linphone.specs=“lime”
User-Agent: LinphoneAndroid/5.1.4 (Galaxy Note20) LinphoneSDK/5.2.110 (tags/5.2.110^0)

v=0
o=6004 115 1059 IN IP4 10.128.1.55
s=Talk
c=IN IP4 10.128.1.55
t=0 0
a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics
a=record:off
m=audio 47929 RTP/AVP 96 97 98 0 8 3 9 99 18 100 101 102 103 104
a=rtpmap:96 opus/48000/2
a=fmtp:96 useinbandfec=1
a=rtpmap:97 speex/16000
a=fmtp:97 vbr=on
a=rtpmap:98 speex/8000
a=fmtp:98 vbr=on
a=rtpmap:99 iLBC/8000
a=fmtp:99 mode=30
a=fmtp:18 annexb=yes
a=rtpmap:100 speex/32000
a=fmtp:100 vbr=on
a=rtpmap:101 telephone-event/48000
a=rtpmap:102 telephone-event/16000
a=rtpmap:103 telephone-event/8000
a=rtpmap:104 telephone-event/32000
a=rtcp:45564
a=rtcp-fb:* trr-int 1000
a=rtcp-fb:* ccm tmmbr

<— Transmitting SIP response (453 bytes) to UDP:10.128.1.55:40340 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.128.1.55:40340;rport=40340;received=10.128.1.55;branch=z9hG4bK.OjyEIFqW8
Call-ID: cCvlGJ2hTN
From: sip:6004@10.128.110.26;tag=mpXmZR2eX
To: sip:7000@10.128.110.26;tag=z9hG4bK.OjyEIFqW8
CSeq: 20 INVITE
WWW-Authenticate: Digest realm=“asterisk”,nonce=“1702444613/491a4560ac37d61a46ae8cd6f31d0786”,opaque=“171437ca0d9bfb65”,algorithm=MD5,qop=“auth”
Server: Asterisk PBX 21.0.0
Content-Length: 0

<— Received SIP request (349 bytes) from UDP:10.128.1.55:40340 —>
ACK sip:7000@10.128.110.26 SIP/2.0
Via: SIP/2.0/UDP 10.128.1.55:40340;branch=z9hG4bK.OjyEIFqW8;rport
Call-ID: cCvlGJ2hTN
From: sip:6004@10.128.110.26;tag=mpXmZR2eX
To: sip:7000@10.128.110.26;tag=z9hG4bK.OjyEIFqW8
Contact: sip:6004@10.128.1.55:40340;transport=udp;expires=3599;+org.linphone.specs=“lime”
Max-Forwards: 70
CSeq: 20 ACK

<— Received SIP request (1589 bytes) from UDP:10.128.1.55:40340 —>
INVITE sip:7000@10.128.110.26 SIP/2.0
Via: SIP/2.0/UDP 10.128.1.55:40340;branch=z9hG4bK.1dCRZ8Rug;rport
From: sip:6004@10.128.110.26;tag=mpXmZR2eX
To: sip:7000@10.128.110.26
CSeq: 21 INVITE
Call-ID: cCvlGJ2hTN
Max-Forwards: 70
Supported: replaces, outbound, gruu, path, record-aware
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, PRACK, UPDATE
Content-Type: application/sdp
Content-Length: 684
Contact: sip:6004@10.128.1.55:40340;transport=udp;expires=3599;+org.linphone.specs=“lime”
User-Agent: LinphoneAndroid/5.1.4 (Galaxy Note20) LinphoneSDK/5.2.110 (tags/5.2.110^0)
Authorization: Digest realm=“asterisk”, nonce=“1702444613/491a4560ac37d61a46ae8cd6f31d0786”, algorithm=MD5, opaque=“171437ca0d9bfb65”, username=“6004”, uri="sip:7000@10.128.110.26", response=“8196ba1705c100034334abc5156f6b9a”, cnonce=“HSfWYcGbrs1UTp4U”, nc=00000001, qop=auth

v=0
o=6004 115 1059 IN IP4 10.128.1.55
s=Talk
c=IN IP4 10.128.1.55
t=0 0
a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics
a=record:off
m=audio 47929 RTP/AVP 96 97 98 0 8 3 9 99 18 100 101 102 103 104
a=rtpmap:96 opus/48000/2
a=fmtp:96 useinbandfec=1
a=rtpmap:97 speex/16000
a=fmtp:97 vbr=on
a=rtpmap:98 speex/8000
a=fmtp:98 vbr=on
a=rtpmap:99 iLBC/8000
a=fmtp:99 mode=30
a=fmtp:18 annexb=yes
a=rtpmap:100 speex/32000
a=fmtp:100 vbr=on
a=rtpmap:101 telephone-event/48000
a=rtpmap:102 telephone-event/16000
a=rtpmap:103 telephone-event/8000
a=rtpmap:104 telephone-event/32000
a=rtcp:45564
a=rtcp-fb:* trr-int 1000
a=rtcp-fb:* ccm tmmbr

<— Transmitting SIP response (279 bytes) to UDP:10.128.1.55:40340 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.128.1.55:40340;rport=40340;received=10.128.1.55;branch=z9hG4bK.1dCRZ8Rug
Call-ID: cCvlGJ2hTN
From: sip:6004@10.128.110.26;tag=mpXmZR2eX
To: sip:7000@10.128.110.26
CSeq: 21 INVITE
Server: Asterisk PBX 21.0.0
Content-Length: 0

-- Executing [7000@from-internal:1] Dial("PJSIP/6004-00000042", "PJSIP/7000@svetah,30") in new stack

<— Transmitting SIP request (904 bytes) to UDP:172.20.1.2:5060 —>
INVITE sip:7000@172.20.1.2 SIP/2.0
Via: SIP/2.0/UDP 10.128.110.26:5060;rport;branch=z9hG4bKPj5cef5cae-40b5-462c-8996-ba099c5cc9f6
From: “IT-Support” sip:6004@10.128.110.26;tag=17f063c8-232a-4745-bd6c-135f4824d487
To: sip:7000@172.20.1.2
Contact: sip:asterisk@10.128.110.26:5060
Call-ID: bdb05d9f-1e6f-4463-8d7b-088df66555ce
CSeq: 18187 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 21.0.0
Content-Type: application/sdp
Content-Length: 237