Pavlidis Savas
2004-Oct-30 01:47 UTC
[Asterisk-Users] HELP: problem making calls from legacy pbx to cisco sip phone via asterisk
I have a peculiar problem. I have installed asterisk and also g729 (2 channels). I have a Cisco7940 IP phone with SIP installed (v6) and a cisco router 2650xm which has an isdn bri voice interface that connects to a legacy pbx system. Also I installed a x-lite to make some tests. I have configured everything after a lot of search and trial and error. So I have managed to make calls from the 7940 to x-lite and vice-versa and also to make calls to to legacy phones from the 7940 or the x-lite via the cisco router using its voice interface. BUT the problem is that from the legacy PBX phones I can call the x-lite but not the cisco 7940 IP Phone. Where is the problem???? Can anyone help me? here are the configurations: SIP.CONF [general] context=default ; Default context for incoming calls port=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all) srvlookup=yes ; Enable DNS SRV lookups on outbound calls disallow=all allow=alaw allow=ulaw allow=gsm allow=g729 [xlite1] type=friend regexten=1239 ; When they register, create extension 1239 username=xlite1 callerid="Savas Pavlidis" <1239> host=dynamic ;nat=yes ; X-Lite is behind a NAT router canreinvite=no ; Typically set to NO if behind NAT [10.1.1.1] ; Cisco 2650XM router type=friend host=10.1.1.1 dtmfmode=rfc2833 disallow=all allow=alaw allow=g729 [419] ; 7940G Cisco IP Phone type=friend username=419 host=dynamic canreinvite=yes dtmfmode=inband disallow=all allow=g729 EXTENSIONS.CONF (PART OF IT) ; The numbers 3XX belong to the traditional ; PBX telephones. ; exten => _3XX,1,Dial(SIP/${EXTEN}@10.1.1.1) exten => _3XX,n,Congestion ; ; ; exten => 419,1,Dial(SIP/419) exten => 420,1,Dial(SIP/xlite1) exten => 420,2,Congestion ; as you may understand 419 is the cisco ip phone ; and extension 420 is the softp phone x-lite ; on the pc. CISCO ROUTER CONFIGURATION (PART OF IT) dial-peer voice 1 pots destination-pattern 3.. direct-inward-dial port 1/0/0 forward-digits all ! dial-peer voice 2 pots destination-pattern 3.. direct-inward-dial port 1/0/1 forward-digits all ! ! dial-peer voice 100 voip destination-pattern 9.. session target ipv4:100.0.0.1 dtmf-relay h245-signal h245-alphanumeric ip qos dscp cs5 media ! dial-peer voice 101 voip destination-pattern 8.. session target ipv4:100.0.0.1 dtmf-relay h245-signal h245-alphanumeric ip qos dscp cs5 media ! dial-peer voice 103 voip destination-pattern 1.. session target ipv4:200.200.201.2 dtmf-relay h245-signal h245-alphanumeric ip qos dscp cs5 media ! dial-peer voice 200 voip destination-pattern 40. session target ipv4:100.0.0.191 dtmf-relay h245-signal h245-alphanumeric ip qos dscp cs5 media ! dial-peer voice 201 voip destination-pattern 5.. session target ipv4:100.0.0.191 dtmf-relay h245-signal h245-alphanumeric ip qos dscp cs5 media ! dial-peer voice 202 voip destination-pattern 42. session protocol sipv2 session target sip-server session transport udp dtmf-relay rtp-nte codec g711alaw no vad ! dial-peer voice 205 voip destination-pattern 41. session protocol sipv2 session target sip-server session transport udp dtmf-relay rtp-nte codec g711alaw no vad ! sip-ua retry invite 3 retry response 3 retry bye 3 retry cancel 3 timers trying 1000 sip-server ipv4:10.1.1.250:5060 ! -------------- next part -------------- A non-text attachment was scrubbed... Name: pavlidis.vcf Type: text/x-vcard Size: 173 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20041030/36b4f93f/pavlidis.vcf
Pavlidis Savas
2004-Nov-01 02:30 UTC
[Asterisk-Users] HELP: problem making calls from legacy pbx to cisco sip phone via asterisk
I have a peculiar problem. I have installed asterisk and also g729 (2 channels). I have a Cisco7940 IP phone with SIP installed (v6) and a cisco router 2650xm which has an isdn bri voice interface that connects to a legacy pbx system. Also I installed a x-lite to make some tests. I have configured everything after a lot of search and trial and error. So I have managed to make calls from the 7940 to x-lite and vice-versa and also to make calls to to legacy phones from the 7940 or the x-lite via the cisco router using its voice interface. BUT the problem is that from the legacy PBX phones I can call the x-lite but not the cisco 7940 IP Phone. I place the call and the cisco phone rings just once (and it shows for a fraction of second the caller id) and then the connections closes as if the called has hunged up. Where is the problem???? Can anyone help me? here are the configurations: SIP.CONF [general] context=default ; Default context for incoming calls port=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all) srvlookup=yes ; Enable DNS SRV lookups on outbound calls disallow=all allow=alaw allow=ulaw allow=gsm allow=g729 [xlite1] type=friend regexten=1239 ; When they register, create extension 1239 username=xlite1 callerid="Savas Pavlidis" <1239> host=dynamic ;nat=yes ; X-Lite is behind a NAT router canreinvite=no ; Typically set to NO if behind NAT [10.1.1.1] ; Cisco 2650XM router type=friend host=10.1.1.1 dtmfmode=rfc2833 disallow=all allow=alaw allow=g729 [419] ; 7940G Cisco IP Phone type=friend username=419 host=dynamic canreinvite=yes dtmfmode=inband disallow=all allow=g729 EXTENSIONS.CONF (PART OF IT) ; The numbers 3XX belong to the traditional ; PBX telephones. ; exten => _3XX,1,Dial(SIP/${EXTEN}@10.1.1.1) exten => _3XX,n,Congestion ; ; ; exten => 419,1,Dial(SIP/419) exten => 420,1,Dial(SIP/xlite1) exten => 420,2,Congestion ; as you may understand 419 is the cisco ip phone ; and extension 420 is the softp phone x-lite ; on the pc. CISCO ROUTER CONFIGURATION (PART OF IT) dial-peer voice 1 pots destination-pattern 3.. direct-inward-dial port 1/0/0 forward-digits all ! dial-peer voice 2 pots destination-pattern 3.. direct-inward-dial port 1/0/1 forward-digits all ! ! dial-peer voice 100 voip destination-pattern 9.. session target ipv4:100.0.0.1 dtmf-relay h245-signal h245-alphanumeric ip qos dscp cs5 media ! dial-peer voice 101 voip destination-pattern 8.. session target ipv4:100.0.0.1 dtmf-relay h245-signal h245-alphanumeric ip qos dscp cs5 media ! dial-peer voice 103 voip destination-pattern 1.. session target ipv4:200.200.201.2 dtmf-relay h245-signal h245-alphanumeric ip qos dscp cs5 media ! dial-peer voice 200 voip destination-pattern 40. session target ipv4:100.0.0.191 dtmf-relay h245-signal h245-alphanumeric ip qos dscp cs5 media ! dial-peer voice 201 voip destination-pattern 5.. session target ipv4:100.0.0.191 dtmf-relay h245-signal h245-alphanumeric ip qos dscp cs5 media ! dial-peer voice 202 voip destination-pattern 42. session protocol sipv2 session target sip-server session transport udp dtmf-relay rtp-nte codec g711alaw no vad ! dial-peer voice 205 voip destination-pattern 41. session protocol sipv2 session target sip-server session transport udp dtmf-relay rtp-nte codec g711alaw no vad ! sip-ua retry invite 3 retry response 3 retry bye 3 retry cancel 3 timers trying 1000 sip-server ipv4:10.1.1.250:5060 ! This is the SIP transaction Sip read: INVITE sip:419@10.1.1.250:5060 SIP/2.0 Via: SIP/2.0/UDP 10.1.1.1:5060 From: <sip:319@10.1.1.1>;tag=8BF5F286-8F5 To: <sip:419@10.1.1.250> Date: Mon, 01 Nov 2004 08:12:53 GMT Call-ID: A9ADD151-2B1411D9-BA7FFFA2-8806CD5A@10.1.1.1 Supported: timer,100rel Min-SE: 1800 Cisco-Guid: 2846660713-722735577-3128754082-2282147162 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER CSeq: 101 INVITE Max-Forwards: 6 Remote-Party-ID: <sip:319@10.1.1.1>;party=calling;screen=no;privacy=off Timestamp: 1099296773 Contact: <sip:319@10.1.1.1:5060> Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Length: 235 v=0 o=CiscoSystemsSIP-GW-UserAgent 1213 9211 IN IP4 10.1.1.1 s=SIP Call c=IN IP4 10.1.1.1 t=0 0 m=audio 18644 RTP/AVP 8 101 c=IN IP4 10.1.1.1 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 20 headers, 11 lines Using latest request as basis request Sending to 10.1.1.1 : 5060 (non-NAT) Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 10.1.1.1:18644 Found description format PCMA Found description format telephone-event Capabilities: us - 0x10e(GSM|ULAW|ALAW|G729A), peer - audio=0x8(ALAW)/video=0x0(EMPTY), combined - 0x8(ALAW) Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 0x1(G723) Found no matching peer or user for '10.1.1.1:56528' Looking for 419 in default list_route: hop: <sip:319@10.1.1.1:5060> Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.1.1.1:5060 From: <sip:319@10.1.1.1>;tag=8BF5F286-8F5 To: <sip:419@10.1.1.250>;tag=as4996755e Call-ID: A9ADD151-2B1411D9-BA7FFFA2-8806CD5A@10.1.1.1 CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:419@10.1.1.250> Content-Length: 0 to 10.1.1.1:5060 We're at 10.1.1.250 port 16084 Answering with preferred capability 0x100(G729A) Answering with non-codec capability 0x1(G723) 12 headers, 10 lines Reliably Transmitting: INVITE sip:419@10.1.1.18:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.1.1.250:5060;branch=z9hG4bK6877ba2f From: "319" <sip:319@10.1.1.250>;tag=as339f0f84 To: <sip:419@10.1.1.18:5060;user=phone> Contact: <sip:319@10.1.1.250> Call-ID: 05785b314473c336521dbfa40a4b8e7e@10.1.1.250 CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Mon, 01 Nov 2004 08:22:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 214 v=0 o=root 28878 28878 IN IP4 10.1.1.250 s=session c=IN IP4 10.1.1.250 t=0 0 m=audio 16084 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - (no NAT) to 10.1.1.18:5060 Sip read: CANCEL sip:419@10.1.1.250:5060 SIP/2.0 Via: SIP/2.0/UDP 10.1.1.1:5060 From: <sip:319@10.1.1.1>;tag=8BF5F286-8F5 To: <sip:419@10.1.1.250> Date: Mon, 01 Nov 2004 08:12:53 GMT Call-ID: A9ADD151-2B1411D9-BA7FFFA2-8806CD5A@10.1.1.1 CSeq: 101 CANCEL Max-Forwards: 6 Timestamp: 1099296773 Content-Length: 0 10 headers, 0 lines Sending to 10.1.1.1 : 5060 (non-NAT) Reliably Transmitting (no NAT): SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 10.1.1.1:5060 From: <sip:319@10.1.1.1>;tag=8BF5F286-8F5 To: <sip:419@10.1.1.250>;tag=as4996755e Call-ID: A9ADD151-2B1411D9-BA7FFFA2-8806CD5A@10.1.1.1 CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:419@10.1.1.250> Content-Length: 0 to 10.1.1.1:5060 Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.1.1:5060 From: <sip:319@10.1.1.1>;tag=8BF5F286-8F5 To: <sip:419@10.1.1.250>;tag=as4996755e Call-ID: A9ADD151-2B1411D9-BA7FFFA2-8806CD5A@10.1.1.1 CSeq: 101 CANCEL User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:419@10.1.1.250> Content-Length: 0 to 10.1.1.1:5060 Reliably Transmitting: CANCEL sip:419@10.1.1.18:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.1.1.250:5060;branch=z9hG4bK6877ba2f From: "319" <sip:319@10.1.1.250>;tag=as339f0f84 To: <sip:419@10.1.1.18:5060;user=phone> Contact: <sip:319@10.1.1.250> Call-ID: 05785b314473c336521dbfa40a4b8e7e@10.1.1.250 CSeq: 102 CANCEL User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 10.1.1.18:5060 Scheduling destruction of call '05785b314473c336521dbfa40a4b8e7e@10.1.1.250' in 15000 ms Sip read: ACK sip:419@10.1.1.250:5060 SIP/2.0 Via: SIP/2.0/UDP 10.1.1.1:5060 From: <sip:319@10.1.1.1>;tag=8BF5F286-8F5 To: <sip:419@10.1.1.250>;tag=as4996755e Date: Mon, 01 Nov 2004 08:12:53 GMT Call-ID: A9ADD151-2B1411D9-BA7FFFA2-8806CD5A@10.1.1.1 Max-Forwards: 6 Content-Length: 0 CSeq: 101 ACK 9 headers, 0 lines Destroying call 'A9ADD151-2B1411D9-BA7FFFA2-8806CD5A@10.1.1.1' Sip read: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.1.1.250:5060;branch=z9hG4bK6877ba2f From: "319" <sip:319@10.1.1.250>;tag=as339f0f84 To: <sip:419@10.1.1.18:5060;user=phone> Call-ID: 05785b314473c336521dbfa40a4b8e7e@10.1.1.250 Date: Mon, 01 Nov 2004 08:12:52 GMT CSeq: 102 INVITE Server: CSCO/6 Contact: <sip:419@10.1.1.18:5060> Content-Length: 0 10 headers, 0 lines Sip read: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.1.1.250:5060;branch=z9hG4bK6877ba2f From: "319" <sip:319@10.1.1.250>;tag=as339f0f84 To: <sip:419@10.1.1.18:5060;user=phone>;tag=000dbc445e500004305d523c-08a4065e Call-ID: 05785b314473c336521dbfa40a4b8e7e@10.1.1.250 Date: Mon, 01 Nov 2004 08:12:52 GMT CSeq: 102 INVITE Server: CSCO/6 Contact: <sip:419@10.1.1.18:5060> Content-Length: 0 10 headers, 0 lines Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.1.250:5060;branch=z9hG4bK6877ba2f From: "319" <sip:319@10.1.1.250>;tag=as339f0f84 To: <sip:419@10.1.1.18:5060;user=phone>;tag=000dbc445e500004305d523c-08a4065e Call-ID: 05785b314473c336521dbfa40a4b8e7e@10.1.1.250 Date: Mon, 01 Nov 2004 08:12:52 GMT CSeq: 102 CANCEL Server: CSCO/6 Content-Length: 0 9 headers, 0 lines Sip read: SIP/2.0 487 Request Cancelled Via: SIP/2.0/UDP 10.1.1.250:5060;branch=z9hG4bK6877ba2f From: "319" <sip:319@10.1.1.250>;tag=as339f0f84 To: <sip:419@10.1.1.18:5060;user=phone>;tag=000dbc445e500004305d523c-08a4065e Call-ID: 05785b314473c336521dbfa40a4b8e7e@10.1.1.250 Date: Mon, 01 Nov 2004 08:12:52 GMT CSeq: 102 INVITE Server: CSCO/6 Contact: <sip:419@10.1.1.18:5060> Content-Length: 0 10 headers, 0 lines Transmitting: ACK sip:419@10.1.1.18:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.1.1.250:5060;branch=z9hG4bK6877ba2f From: "319" <sip:319@10.1.1.250>;tag=as339f0f84 To: <sip:419@10.1.1.18:5060;user=phone>;tag=000dbc445e500004305d523c-08a4065e Contact: <sip:319@10.1.1.250> Call-ID: 05785b314473c336521dbfa40a4b8e7e@10.1.1.250 CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 10.1.1.18:5060 Destroying call '05785b314473c336521dbfa40a4b8e7e@10.1.1.250' sip no debug SIP Debugging Disabled -------------- next part -------------- A non-text attachment was scrubbed... Name: pavlidis.vcf Type: text/x-vcard Size: 173 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20041101/896eddf1/pavlidis.vcf
Pavlidis Savas
2004-Nov-01 05:38 UTC
[Asterisk-Users] HELP: problem making calls from legacy pbx to cisco sip phone via asterisk
Propably this is a cisco router issue. I discovered that if a put play line in the extensions.conf so that it can play something before the call is done, even for one second, the call is working normally. I also played with other variables in the sip.conf but have not succeeded except with the play line in extensions.conf exten => 419,1,Playback(pbx-transfer) exten => 419,2,Dial(SIP/419)
Pavlidis Savas
2004-Nov-01 07:14 UTC
[Asterisk-Users] HELP: problem making calls from legacy pbx to cisco sip phone via asterisk
Thanks for your reply. The cisco router is 2650xm with the following ==================================================================Cisco Internetwork Operating System Software IOS (tm) C2600 Software (C2600-IS-M), Version 12.2(15)ZJ3, EARLY DEPLOYMENT RELE ASE SOFTWARE (fc2) TAC Support: http://www.cisco.com/tac Copyright (c) 1986-2003 by cisco Systems, Inc. Compiled Fri 26-Sep-03 02:05 by eaarmas Image text-base: 0x80008098, data-base: 0x81ADBC0C ROM: System Bootstrap, Version 12.2(8r) [cmong 8r], RELEASE SOFTWARE (fc1) ROM: C2600 Software (C2600-IS-M), Version 12.2(15)ZJ3, EARLY DEPLOYMENT RELEASE SOFTWARE (fc2) Yalko-Thess uptime is 11 weeks, 3 hours, 1 minute System returned to ROM by reload System restarted at 10:52:42 UTC Mon Aug 16 2004 System image file is "flash:c2600-is-mz.122-15.ZJ3.bin" cisco 2650XM (MPC860P) processor (revision 0x200) with 125952K/5120K bytes of me mory. Processor board ID JAE080102SE (2906160658) M860 processor: part number 5, mask 2 Bridging software. X.25 software, Version 3.0.0. Basic Rate ISDN software, Version 1.1. 1 FastEthernet/IEEE 802.3 interface(s) 1 Serial network interface(s) 3 ISDN Basic Rate interface(s) 32K bytes of non-volatile configuration memory. 32768K bytes of processor board System flash (Read/Write) Configuration register is 0x2102 =================================================================it has a VIC 2BRI-ISDN interface (NT/TE) which connects to the PBX which is a Bosch Tenovis Integral 33EW2 EuroISDN. We use already another similar cisco at another office via a WAN link to transfer data and voice channels between the two PBX's (same). I made some changes to experiment with SIP as the hardware already exists. I somewhere red that my version of Cisco IOS cannot handle the INVITE command according to RFC's. You can see a previous mail with the same header which has the configuration and the SIP transaction. I have already got the 12.3(9) and I will flash it in the first occasion and see if it corrects. In the meantime by playing a line of a small audio like transfer, does the trick and fools the cisco and works correctly. Thanks, and if you got any clues by my configs I would greatly appreciate it. Savas Pavlidis Henry Devito wrote:>We need a little more info such as type of PBX, how it connected to >Router type, IOS version, Router config, and at least your extensions.conf. > >I have setup several * using different Cisco routers and PBX's and not had >any problems. I have a lab where I can setup your config with *. > >