Hi there, I was already writing about this problem, and I still didn't managed to fix it. I thought it was a NAT problem, but now without NAT the problem still remains. The error is like this, when I call from pstn, the call goes trough Cisco AS5300, then goes to FS, then to it's final destination - Ip Phone. Now you can answer the phone, but you can't hear anything on any side. And FS returns lots of SIP 415 (unsupported media type) to Cisco AS5300. When you call from Voip phone to the same pstn number it all works well, so I'm starting to be confused more and more. (and in this case there is no 415 messages). I believe cisco is sending something that FS can't understand, in the INVITE's but that's just a wild guess cause I ain't sure what can FS exactly understand very well. Whoever want's I can send him a tcpdump, it's about 7mb tarred, so I can't send it to the list. Thank you all, Josip
Hi, What kind of codec list do you use on as5300? Do you have maybe ACLs defined? Do you use the same dial-peers for incoming and outgoing calls too? What IOS version do you run? Could you share your cisco running-config? Could you send also the SIP messages (without RTP)? Regards, Tamas Josip Djuricic írta:
Hi there,
I was already writing about this problem, and I still didn't managed to fix it. I thought it was a NAT problem, but now without NAT the problem still remains.
The error is like this, when I call from pstn, the call goes trough Cisco AS5300, then goes to FS, then to it's final destination - Ip Phone.
Now you can answer the phone, but you can't hear anything on any side. And FS returns lots of SIP 415 (unsupported media type) to Cisco AS5300.
When you call from Voip phone to the same pstn number it all works well, so I'm starting to be confused more and more. (and in this case there is no 415 messages).
I believe cisco is sending something that FS can't understand, in the INVITE's but that's just a wild guess cause I ain't sure what can FS exactly understand very well.
Whoever want's I can send him a tcpdump, it's about 7mb tarred, so I can't send it to the list.
Thank you all,
Josip
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I can tell you this isn't the problem ... Sofia enables ip6 by default on linux. The AS5300 will spew out 415 errors like crazy. /b On Mar 13, 2008, at 9:21 AM, Tamas wrote:
Hi,
What kind of codec list do you use on as5300? Do you have maybe ACLs defined? Do you use the same dial-peers for incoming and outgoing calls too? What IOS version do you run? Could you share your cisco running-config? Could you send also the SIP messages (without RTP)?
Regards, Tamas
Josip Djuricic írta:
Hi there,
I was already writing about this problem, and I still didn't managed to fix it. I thought it was a NAT problem, but now without NAT the problem still remains.
The error is like this, when I call from pstn, the call goes trough Cisco AS5300, then goes to FS, then to it's final destination - Ip Phone.
Now you can answer the phone, but you can't hear anything on any side. And FS returns lots of SIP 415 (unsupported media type) to Cisco AS5300.
When you call from Voip phone to the same pstn number it all works well, so I'm starting to be confused more and more. (and in this case there is no 415 messages).
I believe cisco is sending something that FS can't understand, in the INVITE's but that's just a wild guess cause I ain't sure what can FS exactly understand very well.
Whoever want's I can send him a tcpdump, it's about 7mb tarred, so I can't send it to the list.
Thank you all,
Josip
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I can tell you that FS is returning 415 errors, not AS. When sofia is inviting everything is well, but when as sends request then the fun starts Brian West wrote:
I can tell you this isn't the problem ... Sofia enables ip6 by default on linux. The AS5300 will spew out 415 errors like crazy.
/b
On Mar 13, 2008, at 9:21 AM, Tamas wrote:
Hi,
What kind of codec list do you use on as5300? Do you have maybe ACLs defined? Do you use the same dial-peers for incoming and outgoing calls too? What IOS version do you run? Could you share your cisco running-config? Could you send also the SIP messages (without RTP)?
Regards, Tamas
Josip Djuricic írta:
Hi there,
I was already writing about this problem, and I still didn't managed to fix it. I thought it was a NAT problem, but now without NAT the problem still remains.
The error is like this, when I call from pstn, the call goes trough Cisco AS5300, then goes to FS, then to it's final destination - Ip Phone.
Now you can answer the phone, but you can't hear anything on any side. And FS returns lots of SIP 415 (unsupported media type) to Cisco AS5300.
When you call from Voip phone to the same pstn number it all works well, so I'm starting to be confused more and more. (and in this case there is no 415 messages).
I believe cisco is sending something that FS can't understand, in the INVITE's but that's just a wild guess cause I ain't sure what can FS exactly understand very well.
Whoever want's I can send him a tcpdump, it's about 7mb tarred, so I can't send it to the list.
Thank you all,
Josip
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The phone is some chinese shit, but it's same with any other model, or softphone. Cisco, polycom, grandstream, linksys, snom.... Brian West wrote:
what kind of phone is this?
/b
On Mar 13, 2008, at 9:53 AM, Josip Djuricic wrote:
I can tell you that FS is returning 415 errors, not AS.
When sofia is inviting everything is well, but when as sends request then the fun starts
Brian West wrote:
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The cisco is sending the SDP as multipart/mixed which is retarded. This isn't supported by FreeSWITCH. /b On Mar 13, 2008, at 9:53 AM, Josip Djuricic wrote:
I can tell you that FS is returning 415 errors, not AS.
When sofia is inviting everything is well, but when as sends request then the fun starts
Brian West wrote:
I can tell you this isn't the problem ... Sofia enables ip6 by default on linux. The AS5300 will spew out 415 errors like crazy.
/b
On Mar 13, 2008, at 9:21 AM, Tamas wrote:
Hi,
What kind of codec list do you use on as5300? Do you have maybe ACLs defined? Do you use the same dial-peers for incoming and outgoing calls too? What IOS version do you run? Could you share your cisco running-config? Could you send also the SIP messages (without RTP)?
Regards, Tamas
Josip Djuricic írta:
Hi there,
I was already writing about this problem, and I still didn't managed to fix it. I thought it was a NAT problem, but now without NAT the problem still remains.
The error is like this, when I call from pstn, the call goes trough Cisco AS5300, then goes to FS, then to it's final destination - Ip Phone.
Now you can answer the phone, but you can't hear anything on any side. And FS returns lots of SIP 415 (unsupported media type) to Cisco AS5300.
When you call from Voip phone to the same pstn number it all works well, so I'm starting to be confused more and more. (and in this case there is no 415 messages).
I believe cisco is sending something that FS can't understand, in the INVITE's but that's just a wild guess cause I ain't sure what can FS exactly understand very well.
Whoever want's I can send him a tcpdump, it's about 7mb tarred, so I can't send it to the list.
Thank you all,
Josip
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So this is what is causing problems, now I need to check if it can be disabled on cisco somehow. There is no way to get pass this with FS? Brian West wrote:
The cisco is sending the SDP as multipart/mixed which is retarded. This isn't supported by FreeSWITCH.
/b
On Mar 13, 2008, at 9:53 AM, Josip Djuricic wrote:
I can tell you that FS is returning 415 errors, not AS.
When sofia is inviting everything is well, but when as sends request then the fun starts
Brian West wrote:
I can tell you this isn't the problem ... Sofia enables ip6 by default on linux. The AS5300 will spew out 415 errors like crazy.
/b
On Mar 13, 2008, at 9:21 AM, Tamas wrote:
Hi,
What kind of codec list do you use on as5300? Do you have maybe ACLs defined? Do you use the same dial-peers for incoming and outgoing calls too? What IOS version do you run? Could you share your cisco running-config? Could you send also the SIP messages (without RTP)?
Regards, Tamas
Josip Djuricic írta:
Hi there,
I was already writing about this problem, and I still didn't managed to fix it. I thought it was a NAT problem, but now without NAT the problem still remains.
The error is like this, when I call from pstn, the call goes trough Cisco AS5300, then goes to FS, then to it's final destination - Ip Phone.
Now you can answer the phone, but you can't hear anything on any side. And FS returns lots of SIP 415 (unsupported media type) to Cisco AS5300.
When you call from Voip phone to the same pstn number it all works well, so I'm starting to be confused more and more. (and in this case there is no 415 messages).
I believe cisco is sending something that FS can't understand, in the INVITE's but that's just a wild guess cause I ain't sure what can FS exactly understand very well.
Whoever want's I can send him a tcpdump, it's about 7mb tarred, so I can't send it to the list.
Thank you all,
Josip
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Hi Brian, With respect, it's not; it's sending a multipart message, of which the SDP (to which it assigns a Content-Type: of application/sdp) is one. RFC3261 says that multipart bodies are OK *if* the receiving end has said that it'll accept them (I've appended the relevant bit); clearly there's a problem here if you're generating the first message, as you don't know what the receiving end will accept. The Cisco sends a multipart body, gets a 415 back with "Accept: application/sdp", and then resends the INVITE with just the SDP as the body. That all looks pretty reasonable. FreeSWITCH then repeats the 415 message - presumably it's expecting it to be ACKed in the same way as a 407, but the Cisco's not doing so. Cheers -- Dave 7.4.1 Message Body Type The Internet media type of the message body MUST be given by the Content-Type header field. If the body has undergone any encoding such as compression, then this MUST be indicated by the Content- Encoding header field; otherwise, Content-Encoding MUST be omitted. If applicable, the character set of the message body is indicated as part of the Content-Type header-field value. The "multipart" MIME type defined in RFC 2046 [11] MAY be used within the body of the message. Implementations that send requests containing multipart message bodies MUST send a session description as a non-multipart message body if the remote implementation requests this through an Accept header field that does not contain multipart. SIP messages MAY contain binary bodies or body parts. When no explicit charset parameter is provided by the sender, media subtypes of the "text" type are defined to have a default charset value of "UTF-8".
The cisco is sending the SDP as multipart/mixed which is retarded. This isn't supported by FreeSWITCH.
/b
On Mar 13, 2008, at 9:53 AM, Josip Djuricic wrote:
I can tell you that FS is returning 415 errors, not AS.
When sofia is inviting everything is well, but when as sends request then the fun starts
Brian West wrote:
I can tell you this isn't the problem ... Sofia enables ip6 by default on linux. The AS5300 will spew out 415 errors like crazy.
/b
On Mar 13, 2008, at 9:21 AM, Tamas wrote:
Hi,
What kind of codec list do you use on as5300? Do you have maybe ACLs defined? Do you use the same dial-peers for incoming and outgoing calls too? What IOS version do you run? Could you share your cisco running-config? Could you send also the SIP messages (without RTP)?
Regards, Tamas
Josip Djuricic írta:
Hi there,
I was already writing about this problem, and I still didn't managed to fix it. I thought it was a NAT problem, but now without NAT the problem still remains.
The error is like this, when I call from pstn, the call goes trough Cisco AS5300, then goes to FS, then to it's final destination - Ip Phone.
Now you can answer the phone, but you can't hear anything on any side. And FS returns lots of SIP 415 (unsupported media type) to Cisco AS5300.
When you call from Voip phone to the same pstn number it all works well, so I'm starting to be confused more and more. (and in this case there is no 415 messages).
I believe cisco is sending something that FS can't understand, in the INVITE's but that's just a wild guess cause I ain't sure what can FS exactly understand very well.
Whoever want's I can send him a tcpdump, it's about 7mb tarred, so I can't send it to the list.
Thank you all,
Josip
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-- David Knell, Director, 3C Limited T: 020 8114 8901 F: 020 3002 7257 M: 001 415 630 3031 http://www.3c.co.uk
I can almost 100% say that sofia is sending an IP6 in the SDP to the Cisco. Can you provide a sip trace for this? /b On Mar 13, 2008, at 9:12 AM, Josip Djuricic wrote:
Hi there,
I was already writing about this problem, and I still didn't managed to fix it. I thought it was a NAT problem, but now without NAT the problem still remains.
The error is like this, when I call from pstn, the call goes trough Cisco AS5300, then goes to FS, then to it's final destination - Ip Phone.
Now you can answer the phone, but you can't hear anything on any side. And FS returns lots of SIP 415 (unsupported media type) to Cisco AS5300.
When you call from Voip phone to the same pstn number it all works well, so I'm starting to be confused more and more. (and in this case there is no 415 messages).
I believe cisco is sending something that FS can't understand, in the INVITE's but that's just a wild guess cause I ain't sure what can FS exactly understand very well.
Whoever want's I can send him a tcpdump, it's about 7mb tarred, so I can't send it to the list.
Thank you all,
Josip
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No problem people, here are all the traces. If you want I can include tcpdump with rtp. To clarify few things, setup is like this. PSTN <-> CISCO AS5300 (10.0.0.19) <-> FS (89.18.32.19) <-> VoipPhone (89.18.42.151) There is no NAT between any of them. I call from PSTN (38513391900) to Voip Phone behind FS (38514001148) Brian West wrote:
I can almost 100% say that sofia is sending an IP6 in the SDP to the Cisco. Can you provide a sip trace for this?
/b
On Mar 13, 2008, at 9:12 AM, Josip Djuricic wrote:
Hi there,
I was already writing about this problem, and I still didn't managed to fix it. I thought it was a NAT problem, but now without NAT the problem still remains.
The error is like this, when I call from pstn, the call goes trough Cisco AS5300, then goes to FS, then to it's final destination - Ip Phone.
Now you can answer the phone, but you can't hear anything on any side. And FS returns lots of SIP 415 (unsupported media type) to Cisco AS5300.
When you call from Voip phone to the same pstn number it all works well, so I'm starting to be confused more and more. (and in this case there is no 415 messages).
I believe cisco is sending something that FS can't understand, in the INVITE's but that's just a wild guess cause I ain't sure what can FS exactly understand very well.
Whoever want's I can send him a tcpdump, it's about 7mb tarred, so I can't send it to the list.
Thank you all,
Josip
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AS5300#show running-config Building configuration... Current configuration : 4335 bytes ! version 12.3 service timestamps debug datetime msec service timestamps log datetime msec no service password-encryption ! hostname AS5300 ! boot-start-marker boot-end-marker ! enable secret 5 $1$TJe7$rTdScSApy2Mm6XnQR4lKs. ! username admin secret 5 $1$mz/t$WUZntaeWKKmi.vFyqvtEZ/ ! ! resource-pool disable ! no aaa new-model ip subnet-zero ! ! ! trunk group Iskra translation-profile incoming from_Iskra translation-profile outgoing to_Iskra ! isdn switch-type primary-net5 ! voice call send-alert voice rtp send-recv ! voice service pots fax protocol pass-through g711alaw ! voice service voip signaling forward unconditional fax protocol pass-through g711alaw sip ! voice class codec 1 codec preference 1 g711alaw ! ! ! ! ! ! ! ! voice translation-rule 10 rule 1 /^3851/ /1/ type any national rule 2 /^385/ // type any national rule 3 /^/ // type any international ! voice translation-rule 20 rule 1 /^/ /#385/ type national international rule 2 /^/ /#/ type international international ! voice translation-rule 21 rule 1 /^#/ // type any international ! ! voice translation-profile from_Iskra translate calling 20 translate called 20 ! voice translation-profile to_Iskra translate calling 10 translate called 10 ! voice translation-profile to_SIP translate calling 21 translate called 21 ! ! ! ! ! controller E1 0 framing NO-CRC4 clock source line primary pri-group timeslots 1-31 ! controller E1 1 framing NO-CRC4 clock source line secondary 1 pri-group timeslots 1-31 ! controller E1 2 framing NO-CRC4 clock source line secondary 2 pri-group timeslots 1-31 ! controller E1 3 framing NO-CRC4 clock source line secondary 3 pri-group timeslots 1-31 ! ! ! ! interface Ethernet0 no ip address shutdown no clns route-cache ! interface Serial0 no ip address clockrate 2015232 no fair-queue no clns route-cache ! interface Serial1 no ip address shutdown clockrate 2015232 no fair-queue no clns route-cache ! interface Serial2 no ip address shutdown clockrate 2015232 no fair-queue no clns route-cache ! interface Serial3 no ip address shutdown clockrate 2015232 no fair-queue no clns route-cache ! interface Serial0:15 no ip address encapsulation ppp trunk-group Iskra isdn switch-type primary-net5 isdn incoming-voice modem 64 isdn T310 60000 isdn send-alerting no clns route-cache ! interface Serial1:15 no ip address encapsulation ppp trunk-group Iskra isdn switch-type primary-net5 isdn incoming-voice modem 64 isdn T310 60000 isdn send-alerting no clns route-cache ! interface Serial2:15 no ip address encapsulation ppp trunk-group Iskra isdn switch-type primary-net5 isdn incoming-voice modem 64 isdn T310 60000 isdn send-alerting no clns route-cache ! interface Serial3:15 no ip address encapsulation ppp trunk-group Iskra isdn switch-type primary-net5 isdn incoming-voice modem 64 isdn T310 60000 isdn send-alerting no clns route-cache ! interface FastEthernet0 no ip address duplex auto speed auto no clns route-cache ! interface FastEthernet0.500 encapsulation dot1Q 500 ip address 10.0.0.19 255.255.252.0 ! ip classless ip route 0.0.0.0 0.0.0.0 10.0.0.1 no ip http server no ip http secure-server ! ! ! ! ! ! voice-port 0:D echo-cancel coverage 32 no comfort-noise cptone SI bearer-cap Speech ! voice-port 1:D echo-cancel coverage 32 no comfort-noise cptone SI bearer-cap Speech ! voice-port 2:D echo-cancel coverage 32 no comfort-noise cptone SI bearer-cap Speech ! voice-port 3:D echo-cancel coverage 32 no comfort-noise cptone SI bearer-cap Speech ! ! ! dial-peer voice 1 voip voice cut-through alert incoming called-number [1-9]T voice-class codec 1 no vad ! dial-peer voice 100 pots trunkgroup Iskra voice cut-through alert destination-pattern [1-9]T progress_ind alert enable 8 supplementary-service pass-through no digit-strip ! dial-peer voice 2 pots voice cut-through alert incoming called-number .T direct-inward-dial ! dial-peer voice 200 voip voice cut-through alert translation-profile outgoing to_SIP destination-pattern #T voice-class codec 1 session protocol sipv2 session target sip-server no vad ! gateway timer receive-rtcp 4 ! sip-ua disable-early-media 180 sip-server ipv4:89.18.32.19:5080 ! ! ! line con 0 login local line aux 0 line vty 0 4 login local ! end AS5300#debug ccsip all This may severely impact system performance. Continue? [confirm]yAll SIP Call tracing is enabled AS5300# AS5300# AS5300# AS5300# AS5300# AS5300# *Mar 13 14:39:39.711: 0x632D8130 : State change from (STATE_NONE, SUBSTATE_NONE) to (STATE_IDLE, SUBSTATE_NONE) *Mar 13 14:39:39.711: adding call id 14E to table *Mar 13 14:39:39.711: Queued event from SIP SPI : SIPSPI_EV_CC_CALL_SETUP *Mar 13 14:39:39.711: CCSIP-SPI-CONTROL: act_idle_call_setup *Mar 13 14:39:39.711: Using Voice Class Codec, tag = 1 *Mar 13 14:39:39.711: act_idle_call_setup: preferred_codec set[0] type :g711alaw bytes: 160 *Mar 13 14:39:39.711: sipSPICopyPeerDataToCCB: From CLI: Modem NSE payload = 100, Passthrough = 0,Modem relay = 0, Gw-Xid = 1 SPRT latency 200, SPRT Retries = 12, Dict Size = 1024 String Len = 32, Compress dir = 3 *Mar 13 14:39:39.711: sipSPICheckRawMsg: have raw msg from ccapi for type: 102 *Mar 13 14:39:39.711: sipSPICanSetFallbackFlag - Local Fallback is not active *Mar 13 14:39:39.715: Queued event from SIP SPI : SIPSPI_EV_CREATE_CONNECTION *Mar 13 14:39:39.715: 0x632D8130 : State change from (STATE_IDLE, SUBSTATE_NONE) to (STATE_IDLE, SUBSTATE_CONNECTING) *Mar 13 14:39:39.715: 0x632D8130 : State change from (STATE_IDLE, SUBSTATE_CONNECTING) to (STATE_IDLE, SUBSTATE_CONNECTING) *Mar 13 14:39:39.715: ****Adding to UAC table. ccb=0x632D8130 key=253738A2-F04211DC-81719B71-D67790FB@10.0.0.19 *Mar 13 14:39:39.715: sipSPIUsetBillingProfile: sipCallId for billing records = 253738A2-F04211DC-81719B71-D67790FB@10.0.0.19 *Mar 13 14:39:39.715: CCSIP-SPI-CONTROL: act_idle_connection_created *Mar 13 14:39:39.715: CCSIP-SPI-CONTROL: act_idle_connection_created: Connid(1) created to 89.18.32.19:5080, local_port 53942 *Mar 13 14:39:39.715: CCSIP-SPI-CONTROL: sipSPIOutgoingCallSDP *Mar 13 14:39:39.715: sipSPISetMediaSrcAddr: media src addr for stream 1 = 10.0.0.19 *Mar 13 14:39:39.715: sipSPIReserveRtpPort: reserved port 16850 for stream 1 *Mar 13 14:39:39.715: convert_codec_bytes_to_ptime: Values :Codec: g711alaw codecbytes :160, ptime: 20 *Mar 13 14:39:39.719: sip_generate_sdp_xcaps_list: Modem Relay and T38 disabled. X-cap not needed *Mar 13 14:39:39.719: CCSIP-SPI-CONTROL: Clock Time Zone is UTC, same as GMT: Using GMT *Mar 13 14:39:39.719: Received Octet3A=0x83 -> Setting ;screen=yes ;privacy=off *Mar 13 14:39:39.719: sipSPIAddLocalContact *Mar 13 14:39:39.719: Queued event from SIP SPI : SIPSPI_EV_SEND_MESSAGE *Mar 13 14:39:39.719: sip_stats_method *Mar 13 14:39:39.719: sipSPIProcessRtpSessions *Mar 13 14:39:39.719: sipSPIAddStream: Adding stream 1 (callid 334) to the VOIP RTP library *Mar 13 14:39:39.719: sipSPISetMediaSrcAddr: media src addr for stream 1 = 10.0.0.19 *Mar 13 14:39:39.719: sipSPIUpdateRtcpSession: for m-line 1 *Mar 13 14:39:39.719: sipSPIUpdateRtcpSession: rtcp_session info laddr = 10.0.0.19, lport = 16850, raddr = 0.0.0.0, rport=0, do_rtcp=FALSE src_callid = 334, dest_callid = -1, stream type = 1 *Mar 13 14:39:39.723: sipSPIUpdateRtcpSession: No rtp session, creating a new one *Mar 13 14:39:39.723: sipSPIAddStream: AddStream in idle state to open a 'recvonly' media session *Mar 13 14:39:39.723: act_idle_connection_created: Transaction active. Facilities will be queued. *Mar 13 14:39:39.723: 0x632D8130 : State change from (STATE_IDLE, SUBSTATE_CONNECTING) to (STATE_SENT_INVITE, SUBSTATE_NONE) *Mar 13 14:39:39.723: HandleUdpSocketWrites - Using new buffer for sip message *Mar 13 14:39:39.723: Sent: INVITE sip:38514001148@89.18.32.19:5080 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.19:5060;x-route-tag="tgrp:Iskra" From: <sip:38513391900@10.0.0.19>;tag=2D49486C-1855 To: <sip:38514001148@89.18.32.19> Date: Thu, 13 Mar 2008 14:39:39 GMT Call-ID: 253738A2-F04211DC-81719B71-D67790FB@10.0.0.19 Supported: timer,100rel Min-SE: 1800 Cisco-Guid: 624255889-4030861788-2171509617-3598160123 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO CSeq: 101 INVITE Max-Forwards: 6 Remote-Party-ID: <sip:38513391900@10.0.0.19>;party=calling;screen=yes;privacy=off Timestamp: 1205419179 Contact: <sip:38513391900@10.0.0.19:5060> Expires: 180 Allow-Events: telephone-event MIME-Version: 1.0 Content-Type: multipart/mixed;boundary=uniqueBoundary Content-Length: 523 --uniqueBoundary Content-Type: application/sdp v=0 o=CiscoSystemsSIP-GW-UserAgent 5947 1401 IN IP4 10.0.0.19 s=SIP Call c=IN IP4 10.0.0.19 t=0 0 m=audio 16850 RTP/AVP 8 c=IN IP4 10.0.0.19 a=rtpmap:8 PCMA/8000 a=ptime:20 --uniqueBoundary Content-Type: application/gtd Content-Disposition: signal;handling=optional IAM, PRN,isdn*,,NET5*, USI,rate,c,3,c,1 USI,lay1,alaw TMR,02 CPN,04,,1,14001148 CGN,04,,1,y,4,13391900 CPC,09 FCI,,,,,,,y, GCI,25356391f04211dc816e9b71d67790fb --uniqueBoundary-- *Mar 13 14:39:39.731: Received: SIP/2.0 415 Unsupported Media Type Via: SIP/2.0/UDP 10.0.0.19:5060;x-route-tag="tgrp:Iskra" From: <sip:38513391900@10.0.0.19>;tag=2D49486C-1855 To: <sip:38514001148@89.18.32.19>;tag=ag2mFXH0NpctD Call-ID: 253738A2-F04211DC-81719B71-D67790FB@10.0.0.19 CSeq: 101 INVITE Timestamp: 1205419179 0.003927 User-Agent: FreeSWITCH-mod_sofia/1.0.pre3-7870 Accept: application/sdp Accept-Encoding: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO Supported: 100rel, precondition, timer Content-Length: 0 *Mar 13 14:39:39.731: HandleUdpSocketReads :Msg enqueued for SPI with IPaddr: 89.18.32.19:5080 *Mar 13 14:39:39.735: CCSIP-SPI-CONTROL: act_sentinvite_new_message *Mar 13 14:39:39.735: CCSIP-SPI-CONTROL: sipSPICheckResponse *Mar 13 14:39:39.735: sip_stats_status_code *Mar 13 14:39:39.735: Roundtrip delay 16 milliseconds for method INVITE *Mar 13 14:39:39.735: Queued event from SIP SPI : SIPSPI_EV_SEND_MESSAGE *Mar 13 14:39:39.735: sip_stats_method *Mar 13 14:39:39.735: ccsip_set_release_source_for_peer:ownCallId[334], src[4] *Mar 13 14:39:39.735: sipSPIHandleUnsupportedMedia: Remote end does not support multipart bodies. Sending only SDP body. *Mar 13 14:39:39.735: RequestCloseConnection: Closing connid 1 Local Port 53942 *Mar 13 14:39:39.735: Queued event from SIP SPI : SIPSPI_EV_CLOSE_CONNECTION *Mar 13 14:39:39.735: Queued event from SIP SPI : SIPSPI_EV_CREATE_CONNECTION *Mar 13 14:39:39.735: 0x632D8130 : State change from (STATE_SENT_INVITE, SUBSTATE_NONE) to (STATE_SENT_INVITE, SUBSTATE_CONNECTING) *Mar 13 14:39:39.735: 0x632D8130 : State change from (STATE_SENT_INVITE, SUBSTATE_CONNECTING) to (STATE_IDLE, SUBSTATE_CONNECTING) *Mar 13 14:39:39.739: Sent: ACK sip:38514001148@89.18.32.19:5080 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.19:5060;x-route-tag="tgrp:Iskra" From: <sip:38513391900@10.0.0.19>;tag=2D49486C-1855 To: <sip:38514001148@89.18.32.19>;tag=ag2mFXH0NpctD Date: Thu, 13 Mar 2008 14:39:39 GMT Call-ID: 253738A2-F04211DC-81719B71-D67790FB@10.0.0.19 Max-Forwards: 6 Content-Length: 0 CSeq: 101 ACK *Mar 13 14:39:39.739: udpsock_close_connect: Socket fd: 1 closed for connid 1 with remote port: 5080 *Mar 13 14:39:39.739: CCSIP-SPI-CONTROL: act_idle_connection_created *Mar 13 14:39:39.739: CCSIP-SPI-CONTROL: act_idle_connection_created: Connid(1) created to 89.18.32.19:5080, local_port 55870 *Mar 13 14:39:39.739: CCSIP-SPI-CONTROL: Clock Time Zone is UTC, same as GMT: Using GMT *Mar 13 14:39:39.739: Received Octet3A=0x83 -> Setting ;screen=yes ;privacy=off *Mar 13 14:39:39.739: sipSPIAddLocalContact *Mar 13 14:39:39.743: Queued event from SIP SPI : SIPSPI_EV_SEND_MESSAGE *Mar 13 14:39:39.743: sip_stats_method *Mar 13 14:39:39.743: sipSPIProcessRtpSessions *Mar 13 14:39:39.743: sipSPIAddStream: Adding stream 1 (callid 334) to the VOIP RTP library *Mar 13 14:39:39.743: sipSPISetMediaSrcAddr: media src addr for stream 1 = 10.0.0.19 *Mar 13 14:39:39.743: sipSPIUpdateRtcpSession: for m-line 1 *Mar 13 14:39:39.743: sipSPIUpdateRtcpSession: rtcp_session info laddr = 10.0.0.19, lport = 16850, raddr = 0.0.0.0, rport=0, do_rtcp=FALSE src_callid = 334, dest_callid = -1, stream type = 1 *Mar 13 14:39:39.743: sipSPIUpdateRtcpSession rtp session already created - update *Mar 13 14:39:39.743: sipSPIAddStream: AddStream in idle state to open a 'recvonly' media session *Mar 13 14:39:39.743: act_idle_connection_created: Transaction active. Facilities will be queued. *Mar 13 14:39:39.743: 0x632D8130 : State change from (STATE_IDLE, SUBSTATE_CONNECTING) to (STATE_SENT_INVITE, SUBSTATE_NONE) *Mar 13 14:39:39.743: Sent: INVITE sip:38514001148@89.18.32.19:5080 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.19:5060;x-route-tag="tgrp:Iskra" From: <sip:38513391900@10.0.0.19>;tag=2D49486C-1855 To: <sip:38514001148@89.18.32.19> Date: Thu, 13 Mar 2008 14:39:39 GMT Call-ID: 253738A2-F04211DC-81719B71-D67790FB@10.0.0.19 Supported: timer,100rel Min-SE: 1800 Cisco-Guid: 624255889-4030861788-2171509617-3598160123 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO CSeq: 102 INVITE Max-Forwards: 6 Remote-Party-ID: <sip:38513391900@10.0.0.19>;party=calling;screen=yes;privacy=off Timestamp: 1205419179 Contact: <sip:38513391900@10.0.0.19:5060> Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Length: 182 v=0 o=CiscoSystemsSIP-GW-UserAgent 5947 1401 IN IP4 10.0.0.19 s=SIP Call c=IN IP4 10.0.0.19 t=0 0 m=audio 16850 RTP/AVP 8 c=IN IP4 10.0.0.19 a=rtpmap:8 PCMA/8000 a=ptime:20 *Mar 13 14:39:39.751: Received: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.0.0.19:5060;x-route-tag="tgrp:Iskra" From: <sip:38513391900@10.0.0.19>;tag=2D49486C-1855 To: <sip:38514001148@89.18.32.19> Call-ID: 253738A2-F04211DC-81719B71-D67790FB@10.0.0.19 CSeq: 102 INVITE Timestamp: 1205419179 0.007713 User-Agent: FreeSWITCH-mod_sofia/1.0.pre3-7870 Content-Length: 0 *Mar 13 14:39:39.755: HandleUdpSocketReads :Msg enqueued for SPI with IPaddr: 89.18.32.19:5080 *Mar 13 14:39:39.755: CCSIP-SPI-CONTROL: act_sentinvite_new_message *Mar 13 14:39:39.755: CCSIP-SPI-CONTROL: sipSPICheckResponse *Mar 13 14:39:39.755: sip_stats_status_code *Mar 13 14:39:39.755: Roundtrip delay 16 milliseconds for method INVITE *Mar 13 14:39:39.755: 0x632D8130 : State change from (STATE_SENT_INVITE, SUBSTATE_NONE) to (STATE_RECD_PROCEEDING, SUBSTATE_PROCEEDING_PROCEEDING) *Mar 13 14:39:40.499: Received: SIP/2.0 415 Unsupported Media Type Via: SIP/2.0/UDP 10.0.0.19:5060;x-route-tag="tgrp:Iskra" From: <sip:38513391900@10.0.0.19>;tag=2D49486C-1855 To: <sip:38514001148@89.18.32.19>;tag=ag2mFXH0NpctD Call-ID: 253738A2-F04211DC-81719B71-D67790FB@10.0.0.19 CSeq: 101 INVITE Timestamp: 1205419179 0.003927 User-Agent: FreeSWITCH-mod_sofia/1.0.pre3-7870 Accept: application/sdp Accept-Encoding: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO Supported: 100rel, precondition, timer Content-Length: 0 *Mar 13 14:39:40.499: HandleUdpSocketReads :Msg enqueued for SPI with IPaddr: 89.18.32.19:5080 *Mar 13 14:39:40.499: CCSIP-SPI-CONTROL: act_recdproc_new_message *Mar 13 14:39:40.499: CCSIP-SPI-CONTROL: act_recdproc_new_message_response *Mar 13 14:39:40.499: CCSIP-SPI-CONTROL: sipSPICheckResponse *Mar 13 14:39:40.499: CCSIP-SPI-CONTROL: sipSPICheckResponse : Bad cseq number *Mar 13 14:39:40.499: CCSIP-SPI-CONTROL: act_recdproc_new_message_response : Check Response Failed *Mar 13 14:39:40.663: Received: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.0.0.19:5060;x-route-tag="tgrp:Iskra" From: <sip:38513391900@10.0.0.19>;tag=2D49486C-1855 To: <sip:38514001148@89.18.32.19>;tag=BSUDHr23jZ2cS Call-ID: 253738A2-F04211DC-81719B71-D67790FB@10.0.0.19 CSeq: 102 INVITE Contact: <sip:mod_sofia@89.18.32.19:5080;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.0.pre3-7870 Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO Supported: 100rel, precondition, timer Content-Length: 0 *Mar 13 14:39:40.663: HandleUdpSocketReads :Msg enqueued for SPI with IPaddr: 89.18.32.19:5080 *Mar 13 14:39:40.663: CCSIP-SPI-CONTROL: act_recdproc_new_message *Mar 13 14:39:40.663: CCSIP-SPI-CONTROL: act_recdproc_new_message_response *Mar 13 14:39:40.663: CCSIP-SPI-CONTROL: sipSPICheckResponse *Mar 13 14:39:40.663: sip_stats_status_code *Mar 13 14:39:40.663: Roundtrip delay 924 milliseconds for method INVITE *Mar 13 14:39:40.663: CCSIP-SPI-CONTROL: act_recdproc_new_message_response : Early media disabled for 180: Ignoring SDP if present *Mar 13 14:39:40.663: HandleSIP1xxRinging: SDP in 180 will be ignored if present: No early media cut through *Mar 13 14:39:40.663: HandleSIP1xxRinging: SDP Body either absent or ignored in 180 RINGING:- would wait for 200 OK to do negotiation. *Mar 13 14:39:40.663: HandleSIP1xxRinging: MediaNegotiation expected in 200 OK *Mar 13 14:39:40.667: sipSPICreateRawMsg: No GTD passed. *Mar 13 14:39:40.667: 0x632D8130 : State change from (STATE_RECD_PROCEEDING, SUBSTATE_PROCEEDING_PROCEEDING) to (STATE_RECD_PROCEEDING, SUBSTATE_PROCEEDING_ALERTING) *Mar 13 14:39:40.667: HandleSIP1xxRinging: Transaction Complete. Lock on Facilities released. *Mar 13 14:39:40.667: CCSIP-SPI-CONTROL: ccsip_bridge: confID = 121, srcCallID = 334, dstCallID = 333 *Mar 13 14:39:40.667: sipSPIUupdateCcCallIds: old src/dest ccCallids: -1/-1, new src/dest ccCallids: 334/333 *Mar 13 14:39:40.667: sipSPIUupdateCcCallIds: old streamcallid=334, new streamcallid=334 *Mar 13 14:39:40.667: CCSIP-SPI-CONTROL: ccsip_caps_ind *Mar 13 14:39:40.667: ccsip_caps_ind: Load DSP with preferred codec : g711alaw, Bytes=160 *Mar 13 14:39:40.667: ccsip_caps_ind: set forking flag to 0x0 *Mar 13 14:39:40.667: sip_set_modem_caps: Preferred (or the one that came from DSM) modem relay=1, from CLI config=0 *Mar 13 14:39:40.667: sip_set_modem_caps: Disabling Modem Relay... *Mar 13 14:39:40.667: sip_set_modem_caps: Negotiation NOT yet Done. Set the configured values *Mar 13 14:39:40.667: sip_set_modem_caps: Modem Relay & Passthru both disabled *Mar 13 14:39:40.667: sip_set_modem_caps: nse payload = 100, ptru mode = 0, ptru-codec=0, redundancy=0, xid=1, relay=0, sprt-retry=12, latecncy=200, compres-dir=3, dict=1024, strnlen=32 *Mar 13 14:39:40.667: sipSPISetStreamInfo: num_streams = 1 *Mar 13 14:39:40.667: sipSPISetStreamInfo: adding stream type 1 from mline 1 *Mar 13 14:39:40.667: sipSPISetStreamInfo: caps.stream_count=1, caps.stream[0].stream_type=0x2, caps.stream_list.xmitFunc=voip_rtp_xmit, caps.stream_list.context=0x63495DA8 (gccb) *Mar 13 14:39:40.671: ccsip_caps_ind: Load DSP with codec : g711alaw, Bytes=160 *Mar 13 14:39:40.671: CCSIP-SPI-CONTROL: ccsip_caps_ack *Mar 13 14:39:40.671: ccsip_caps_ack: set forking flag to 0x7 *Mar 13 14:39:41.207: Received: SIP/2.0 415 Unsupported Media Type Via: SIP/2.0/UDP 10.0.0.19:5060;x-route-tag="tgrp:Iskra" From: <sip:38513391900@10.0.0.19>;tag=2D49486C-1855 To: <sip:38514001148@89.18.32.19>;tag=ag2mFXH0NpctD Call-ID: 253738A2-F04211DC-81719B71-D67790FB@10.0.0.19 CSeq: 101 INVITE Timestamp: 1205419179 0.003927 User-Agent: FreeSWITCH-mod_sofia/1.0.pre3-7870 Accept: application/sdp Accept-Encoding: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO Supported: 100rel, precondition, timer Content-Length: 0 *Mar 13 14:39:41.207: HandleUdpSocketReads :Msg enqueued for SPI with IPaddr: 89.18.32.19:5080 *Mar 13 14:39:41.207: CCSIP-SPI-CONTROL: act_recdproc_new_message *Mar 13 14:39:41.207: CCSIP-SPI-CONTROL: act_recdproc_new_message_response *Mar 13 14:39:41.207: CCSIP-SPI-CONTROL: sipSPICheckResponse *Mar 13 14:39:41.207: CCSIP-SPI-CONTROL: sipSPICheckResponse : Bad cseq number *Mar 13 14:39:41.207: CCSIP-SPI-CONTROL: act_recdproc_new_message_response : Check Response Failed *Mar 13 14:39:43.711: Received: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.19:5060;x-route-tag="tgrp:Iskra" From: <sip:38513391900@10.0.0.19>;tag=2D49486C-1855 To: <sip:38514001148@89.18.32.19>;tag=BSUDHr23jZ2cS Call-ID: 253738A2-F04211DC-81719B71-D67790FB@10.0.0.19 CSeq: 102 INVITE Contact: <sip:mod_sofia@89.18.32.19:5080;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.0.pre3-7870 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO Supported: 100rel, precondition, timer Min-SE: 1800 Content-Type: application/sdp Content-Disposition: session Content-Length: 272 v=0 o=FreeSWITCH 8367682047008151068 7117515538434833209 IN IP4 89.18.32.19 s=FreeSWITCH c=IN IP4 89.18.32.19 t=0 0 a=sendrecv m=audio 16614 RTP/AVP 8 101 13 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20 *Mar 13 14:39:43.711: HandleUdpSocketReads :Msg enqueued for SPI with IPaddr: 89.18.32.19:5080 *Mar 13 14:39:43.711: ***creating child CCB 0 *Mar 13 14:39:43.711: ***Updating child CCB *Mar 13 14:39:43.711: adding call id 14F to table *Mar 13 14:39:43.711: CCSIP-SPI-CONTROL: act_recdproc_new_message *Mar 13 14:39:43.711: CCSIP-SPI-CONTROL: act_recdproc_new_message_response *Mar 13 14:39:43.711: CCSIP-SPI-CONTROL: sipSPICheckResponse *Mar 13 14:39:43.715: sip_stats_status_code *Mar 13 14:39:43.715: Roundtrip delay 759781384 milliseconds for method INVITE *Mar 13 14:39:43.715: act_recdproc_new_message_response: Transaction active. Facilities will be queued. *Mar 13 14:39:43.715: CCSIP-SPI-CONTROL: sipSPIUACSessionTimer *Mar 13 14:39:43.715: sipSPIGetGtdBody: No valid GTD body found. *Mar 13 14:39:43.715: CCSIP-SPI-CONTROL: act_recdproc_continue_200_processing *Mar 13 14:39:43.715: CCSIP-SPI-CONTROL: act_recdproc_continue_200_processing: *** This ccb is a child *Mar 13 14:39:43.715: Removing call id 14E *Mar 13 14:39:43.715: sipSPIUupdateCcCallIds: old src/dest ccCallids: -1/333, new src/dest ccCallids: 334/333 *Mar 13 14:39:43.715: sipSPIUupdateCcCallIds: old streamcallid=334, new streamcallid=334 *Mar 13 14:39:43.715: sipSPIMakeChildParentCCB: skipping assignment of context ptr *Mar 13 14:39:43.715: ****Deleting from UAC table. ccb=0x632D8130 key=253738A2-F04211DC-81719B71-D67790FB@10.0.0.19 *Mar 13 14:39:43.715: CCSIP-SPI-CONTROL: sipSPICallCleanup *Mar 13 14:39:43.715: ***freeing parent ccb 632D8130 *Mar 13 14:39:43.715: Removing call id 14E *Mar 13 14:39:43.715: sipSPIUdeleteccCallIdFromTable: Entry not found for ccCallId *Mar 13 14:39:43.715: sipSPIFlushEventBufferQueue: There are 0 events on the internal queue that are going to be free'd *Mar 13 14:39:43.715: freeing ccb 632D8130 *Mar 13 14:39:43.715: adding call id 14E to table *Mar 13 14:39:43.715: ****Adding to UAC table. ccb=0x632D909C key=253738A2-F04211DC-81719B71-D67790FB@10.0.0.19 *Mar 13 14:39:43.719: sipSPICompareRespMediaInfo *Mar 13 14:39:43.719: sipSPICompareRespMediaInfo No Comparsion needed as 18x response SDP is either absent or ignored *Mar 13 14:39:43.719: sipSPIDoMediaNegotiation: number of m lines is 1 *Mar 13 14:39:43.719: sipSPIDoAudioNegotiation: Codec (g711alaw) Negotiation Successful on Static Payload for m-line 1 *Mar 13 14:39:43.719: sipSPIDoPtimeNegotiation: One ptime attribute found - value:20 *Mar 13 14:39:43.719: convert_ptime_to_codec_bytes: Values :Codec: g711alaw ptime :20, codecbytes: 160 *Mar 13 14:39:43.719: convert_codec_bytes_to_ptime: Values :Codec: g711alaw codecbytes :160, ptime: 20 *Mar 13 14:39:43.719: sipSPIDoDTMFRelayNegotiation: m-line index 1 *Mar 13 14:39:43.719: sipSPIDoDTMFRelayNegotiation: Requested DTMF-RELAY option(s) not found in Preferred DTMF-RELAY option list! *Mar 13 14:39:43.719: sipSPIStreamTypeAndDtmfRelay: DTMF Relay mode : Inband Voice *Mar 13 14:39:43.719: sip_sdp_get_modem_relay_cap_params: *Mar 13 14:39:43.719: sip_sdp_get_modem_relay_cap_params: NSE payload from X-cap = 0 *Mar 13 14:39:43.719: sip_do_nse_negotiation: SDP not present. Use local NSE payload 100. *Mar 13 14:39:43.719: sip_select_modem_relay_params: X-tmr not present in SDP. Disable modem relay *Mar 13 14:39:43.719: sipSPIGetSDPDirectionAttribute: No direction attribute present or multiple direction attributes that can't be handled for m-line:1 and num-a-lines:0 *Mar 13 14:39:43.719: sipSPIDoAudioNegotiation: Codec negotiation successful for media line 1 payload_type=8, codec_bytes=160, codec=g711alaw, dtmf_relay=inband-voice stream_type=voice-only (0), dest_ip_address=89.18.32.19, dest_port=16614 *Mar 13 14:39:43.719: sipSPIReplaceSDP *Mar 13 14:39:43.719: sipSPICopySdpInfo *Mar 13 14:39:43.719: sipSPIUpdCallWithSdpInfo: Preferred Codec : g711alaw, bytes :160 Preferred DTMF relay : inband-voice Preferred NTE payload : 101 Early Media : No Delayed Media : No Bridge Done : Yes New Media : No DSP DNLD Reqd : Yes *Mar 13 14:39:43.719: sipSPISetMediaSrcAddr: media src addr for stream 1 = 10.0.0.19 *Mar 13 14:39:43.719: sipSPIUpdCallWithSdpInfo:Stream Type:0 M-line Index : 1 State : STREAM_ADDING (2) Callid : 334 Negotiated Codec : g711alaw, bytes :160 Negotiated DTMF relay : inband-voice Negotiated NTE payload : 0 Negotiated CN payload : 0 Media Srce Addr/Port : 10.0.0.19:16850 Media Dest Addr/Port : 89.18.32.19:16614 *Mar 13 14:39:43.719: sipSPIProcessMediaChanges *Mar 13 14:39:43.719: sipSPIProcessMediaChanges DSPDNLD Reqd is true. Call DoCapsInd *Mar 13 14:39:43.723: CCSIP-SPI-CONTROL: ccsip_do_caps_ind *Mar 13 14:39:43.723: ccsip_get_rtcp_session_parameters: CURRENT VALUES: stream_callid=334, current_seq_num=0x119 *Mar 13 14:39:43.723: ccsip_get_rtcp_session_parameters: NEW VALUES: stream_callid=334, current_seq_num=0xCC1 *Mar 13 14:39:43.723: ccsip_do_caps_ind: Load DSP with codec: g711alaw, Bytes=160 *Mar 13 14:39:43.723: sipSPISetStreamInfo: num_streams = 0 *Mar 13 14:39:43.723: sipSPISetStreamInfo: ERROR - number of active streams is zero (0)! *Mar 13 14:39:43.723: sipSPISetStreamInfo: caps.stream_count=0, caps.stream[0].stream_type=0xFFFF, caps.stream_list.xmitFunc=??unknown??, caps.stream_list.context=0x0 (rtp_fork_id) *Mar 13 14:39:43.723: sipSPISetDTMFRelayMode: set DSP for dtmf-relay = CC_CAP_DTMF_RELAY_INBAND_VOICE_AND_OOB *Mar 13 14:39:43.723: sip_set_modem_caps: Preferred (or the one that came from DSM) modem relay=0, from CLI config=0 *Mar 13 14:39:43.723: sip_set_modem_caps: Disabling Modem Relay... *Mar 13 14:39:43.723: sip_set_modem_caps: Negotiation already Done. Set negotiated Modem caps *Mar 13 14:39:43.723: sip_set_modem_caps: Modem Relay & Passthru both disabled *Mar 13 14:39:43.723: sip_set_modem_caps: nse payload = 100, ptru mode = 0, ptru-codec=0, redundancy=0, xid=0, relay=0, sprt-retry=12, latecncy=200, compres-dir=3, dict=1024, strnlen=32 *Mar 13 14:39:43.723: ccsip_process_response_contact_record_route *Mar 13 14:39:43.723: 0x632D909C : State change from (STATE_RECD_PROCEEDING, SUBSTATE_NONE) to (STATE_RECD_PROCEEDING, SUBSTATE_CONNECTING) *Mar 13 14:39:43.723: Queued event from SIP SPI : SIPSPI_EV_CREATE_CONNECTION *Mar 13 14:39:43.723: 0x632D909C : State change from (STATE_RECD_PROCEEDING, SUBSTATE_CONNECTING) to (STATE_RECD_PROCEEDING, SUBSTATE_CONNECTING) *Mar 13 14:39:43.723: CCSIP-SPI-CONTROL: ccsip_caps_ack *Mar 13 14:39:43.723: ccsip_caps_ack: set forking flag to 0x7 *Mar 13 14:39:43.727: CCSIP-SPI-CONTROL: act_recdproc_connection_created *Mar 13 14:39:43.727: CCSIP-SPI-CONTROL: sipSPICheckSocketConnection: Connid(2) created to 89.18.32.19:5080, local_port 58463 *Mar 13 14:39:43.727: 0x632D909C : State change from (STATE_RECD_PROCEEDING, SUBSTATE_CONNECTING) to (STATE_RECD_PROCEEDING, SUBSTATE_NONE) *Mar 13 14:39:43.727: CCSIP-SPI-CONTROL: sipSPIProcess200OKforinvite *Mar 13 14:39:43.727: sipSPICreateRawMsg: No GTD passed. *Mar 13 14:39:43.727: RequestCloseConnection: Closing connid 1 Local Port 0 *Mar 13 14:39:43.727: Queued event from SIP SPI : SIPSPI_EV_CLOSE_CONNECTION *Mar 13 14:39:43.727: Queued event from SIP SPI : SIPSPI_EV_SEND_MESSAGE *Mar 13 14:39:43.727: sip_stats_method *Mar 13 14:39:43.727: 0x632D909C : State change from (STATE_RECD_PROCEEDING, SUBSTATE_NONE) to (STATE_ACTIVE, SUBSTATE_NONE) *Mar 13 14:39:43.727: The Call Setup Information is : Call Control Block (CCB) : 0x632D909C State of The Call : STATE_ACTIVE TCP Sockets Used : NO Calling Number : 38513391900 Called Number : 38514001148 Number of Media Streams : 1 *Mar 13 14:39:43.727: Media Stream 1 Negotiated Codec : g711alaw Negotiated Codec Bytes : 160 Negotiated Dtmf-relay : 0 Dtmf-relay Payload : 0 Source IP Address (Media): 10.0.0.19 Source IP Port (Media): 16850 Destn IP Address (Media): 89.18.32.19 Destn IP Port (Media): 16614 *Mar 13 14:39:43.727: Orig Destn IP Address:Port (Media): 0.0.0.0:0 *Mar 13 14:39:43.731: Source IP Address (Sig ): 10.0.0.19 Destn SIP Req Addr:Port : 89.18.32.19:5080 Destn SIP Resp Addr:Port : 0.0.0.0:0 Destination Name : *Mar 13 14:39:43.731: sipSPIProcess200OKforinvite: Transaction Complete. Lock on Facilities released. *Mar 13 14:39:43.731: sipSPIFlushEventBufferQueue: There are 0 events on the internal queue that are going to be free'd *Mar 13 14:39:43.731: udpsock_close_connect: Socket fd: 1 closed for connid 1 with remote port: 5080 *Mar 13 14:39:43.731: Sent: ACK sip:mod_sofia@89.18.32.19:5080;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.0.0.19:5060 From: <sip:38513391900@10.0.0.19>;tag=2D49486C-1855 To: <sip:38514001148@89.18.32.19>;tag=BSUDHr23jZ2cS Date: Thu, 13 Mar 2008 14:39:39 GMT Call-ID: 253738A2-F04211DC-81719B71-D67790FB@10.0.0.19 Max-Forwards: 6 Content-Length: 0 CSeq: 102 ACK *Mar 13 14:39:44.591: Received: SIP/2.0 415 Unsupported Media Type Via: SIP/2.0/UDP 10.0.0.19:5060;x-route-tag="tgrp:Iskra" From: <sip:38513391900@10.0.0.19>;tag=2D49486C-1855 To: <sip:38514001148@89.18.32.19>;tag=ag2mFXH0NpctD Call-ID: 253738A2-F04211DC-81719B71-D67790FB@10.0.0.19 CSeq: 101 INVITE Timestamp: 1205419179 0.003927 User-Agent: FreeSWITCH-mod_sofia/1.0.pre3-7870 Accept: application/sdp Accept-Encoding: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO Supported: 100rel, precondition, timer Content-Length: 0 *Mar 13 14:39:44.591: HandleUdpSocketReads :Msg enqueued for SPI with IPaddr: 89.18.32.19:5080 *Mar 13 14:39:44.595: CCSIP-SPI-CONTROL: sipSPICheckFromToResponse: Compare From/To failed - IGNORE IF HAIRPIN CALL *Mar 13 14:39:44.595: ccsip_spi_process_event:Could not find ccb for response *Mar 13 14:39:48.171: Received: SIP/2.0 415 Unsupported Media Type Via: SIP/2.0/UDP 10.0.0.19:5060;x-route-tag="tgrp:Iskra" From: <sip:38513391900@10.0.0.19>;tag=2D49486C-1855 To: <sip:38514001148@89.18.32.19>;tag=ag2mFXH0NpctD Call-ID: 253738A2-F04211DC-81719B71-D67790FB@10.0.0.19 CSeq: 101 INVITE Timestamp: 1205419179 0.003927 User-Agent: FreeSWITCH-mod_sofia/1.0.pre3-7870 Accept: application/sdp Accept-Encoding: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO Supported: 100rel, precondition, timer Content-Length: 0 *Mar 13 14:39:48.171: HandleUdpSocketReads :Msg enqueued for SPI with IPaddr: 89.18.32.19:5080 *Mar 13 14:39:48.171: CCSIP-SPI-CONTROL: sipSPICheckFromToResponse: Compare From/To failed - IGNORE IF HAIRPIN CALL *Mar 13 14:39:48.171: ccsip_spi_process_event:Could not find ccb for response *Mar 13 14:39:49.747: %ISDN-6-CONNECT: Interface Serial1:30 is now connected to 13391900 unknown *Mar 13 14:39:50.319: Received: SIP/2.0 415 Unsupported Media Type Via: SIP/2.0/UDP 10.0.0.19:5060;x-route-tag="tgrp:Iskra" From: <sip:38513391900@10.0.0.19>;tag=2D49486C-1855 To: <sip:38514001148@89.18.32.19>;tag=ag2mFXH0NpctD Call-ID: 253738A2-F04211DC-81719B71-D67790FB@10.0.0.19 CSeq: 101 INVITE Timestamp: 1205419179 0.003927 User-Agent: FreeSWITCH-mod_sofia/1.0.pre3-7870 Accept: application/sdp Accept-Encoding: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO Supported: 100rel, precondition, timer Content-Length: 0 *Mar 13 14:39:50.323: HandleUdpSocketReads :Msg enqueued for SPI with IPaddr: 89.18.32.19:5080 *Mar 13 14:39:50.323: CCSIP-SPI-CONTROL: sipSPICheckFromToResponse: Compare From/To failed - IGNORE IF HAIRPIN CALL *Mar 13 14:39:50.323: ccsip_spi_process_event:Could not find ccb for response *Mar 13 14:39:50.719: Received: BYE sip:38513391900@10.0.0.19:5060 SIP/2.0 Via: SIP/2.0/UDP 89.18.32.19:5080;rport;branch=z9hG4bK71yv201UjDa8K Max-Forwards: 70 From: <sip:38514001148@89.18.32.19>;tag=BSUDHr23jZ2cS To: <sip:38513391900@10.0.0.19>;tag=2D49486C-1855 Call-ID: 253738A2-F04211DC-81719B71-D67790FB@10.0.0.19 CSeq: 96584005 BYE Contact: <sip:mod_sofia@89.18.32.19:5080;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.0.pre3-7870 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO Supported: 100rel, precondition, timer Reason: Q.850;cause=16;text="NORMAL_CLEARING" Content-Length: 0 *Mar 13 14:39:50.719: HandleUdpSocketReads :Msg enqueued for SPI with IPaddr: 89.18.32.19:5080 *Mar 13 14:39:50.719: ****Found CCB in UAC table *Mar 13 14:39:50.719: CCSIP-SPI-CONTROL: act_active_new_message *Mar 13 14:39:50.719: CCSIP-SPI-CONTROL: sact_active_new_message_request *Mar 13 14:39:50.719: CCSIP-SPI-CONTROL: Clock Time Zone is UTC, same as GMT: Using GMT *Mar 13 14:39:50.719: sip_stats_method *Mar 13 14:39:50.719: ccsip_set_release_source_for_peer:ownCallId[334], src[4] *Mar 13 14:39:50.719: sipSPIStopHoldTimer: Stopping hold timer *Mar 13 14:39:50.723: Queued event from SIP SPI : SIPSPI_EV_SEND_MESSAGE *Mar 13 14:39:50.723: sip_stats_status_code *Mar 13 14:39:50.723: CCSIP-SPI-CONTROL: sipSPIInitiateCallDisconnect : Initiate call disconnect(16) for outgoing call *Mar 13 14:39:50.723: 0x632D909C : State change from (STATE_ACTIVE, SUBSTATE_NONE) to (STATE_DISCONNECTING, SUBSTATE_NONE) *Mar 13 14:39:50.723: Sent: SIP/2.0 200 OK Via: SIP/2.0/UDP 89.18.32.19:5080;rport;branch=z9hG4bK71yv201UjDa8K From: <sip:38514001148@89.18.32.19>;tag=BSUDHr23jZ2cS To: <sip:38513391900@10.0.0.19>;tag=2D49486C-1855 Date: Thu, 13 Mar 2008 14:39:50 GMT Call-ID: 253738A2-F04211DC-81719B71-D67790FB@10.0.0.19 Server: Cisco-SIPGateway/IOS-12.x Content-Length: 0 CSeq: 96584005 BYE *Mar 13 14:39:50.723: Queued event from SIP SPI : SIPSPI_EV_CC_CALL_DISCONNECT *Mar 13 14:39:50.727: CCSIP-SPI-CONTROL: act_disconnecting_disconnect *Mar 13 14:39:50.727: CCSIP-SPI-CONTROL: sipSPICallCleanup *Mar 13 14:39:50.727: sipSPIIcpifUpdate :CallState: 4 Playout: 0 DiscTime:75978839 ConnTime 75978139 *Mar 13 14:39:50.727: 0x632D909C : State change from (STATE_DISCONNECTING, SUBSTATE_NONE) to (STATE_DEAD, SUBSTATE_NONE) *Mar 13 14:39:50.727: The Call Setup Information is : Call Control Block (CCB) : 0x632D909C State of The Call : STATE_DEAD TCP Sockets Used : NO Calling Number : 38513391900 Called Number : 38514001148 Number of Media Streams : 1 *Mar 13 14:39:50.727: Media Stream 1 Negotiated Codec : g711alaw Negotiated Codec Bytes : 160 Negotiated Dtmf-relay : 0 Dtmf-relay Payload : 0 Source IP Address (Media): 10.0.0.19 Source IP Port (Media): 16850 Destn IP Address (Media): 89.18.32.19 Destn IP Port (Media): 16614 *Mar 13 14:39:50.727: Orig Destn IP Address:Port (Media): 0.0.0.0:0 *Mar 13 14:39:50.727: Source IP Address (Sig ): 10.0.0.19 Destn SIP Req Addr:Port : 89.18.32.19:5080 Destn SIP Resp Addr:Port : 89.18.32.19:5080 Destination Name : *Mar 13 14:39:50.727: Disconnect Cause (CC) : 16 Disconnect Cause (SIP) : 200 *Mar 13 14:39:50.727: ****Deleting from UAC table. ccb=0x632D909C key=253738A2-F04211DC-81719B71-D67790FB@10.0.0.19 *Mar 13 14:39:50.727: Removing call id 14E *Mar 13 14:39:50.731: Removing call id 14F *Mar 13 14:39:50.731: RequestCloseConnection: Closing connid 2 Local Port 58463 *Mar 13 14:39:50.731: Queued event from SIP SPI : SIPSPI_EV_CLOSE_CONNECTION *Mar 13 14:39:50.731: sipSPIFlushEventBufferQueue: There are 0 events on the internal queue that are going to be free'd *Mar 13 14:39:50.731: freeing ccb 632D909C *Mar 13 14:39:50.731: udpsock_close_connect: Socket fd: 2 closed for connid 2 with remote port: 5080 *Mar 13 14:39:50.735: %ISDN-6-DISCONNECT: Interface Serial1:30 disconnected from 13391900 , call lasted 6 seconds *Mar 13 14:39:55.279: Received: SIP/2.0 415 Unsupported Media Type Via: SIP/2.0/UDP 10.0.0.19:5060;x-route-tag="tgrp:Iskra" From: <sip:38513391900@10.0.0.19>;tag=2D49486C-1855 To: <sip:38514001148@89.18.32.19>;tag=ag2mFXH0NpctD Call-ID: 253738A2-F04211DC-81719B71-D67790FB@10.0.0.19 CSeq: 101 INVITE Timestamp: 1205419179 0.003927 User-Agent: FreeSWITCH-mod_sofia/1.0.pre3-7870 Accept: application/sdp Accept-Encoding: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO Supported: 100rel, precondition, timer Content-Length: 0 *Mar 13 14:39:55.279: HandleUdpSocketReads :Msg enqueued for SPI with IPaddr: 89.18.32.19:5080 *Mar 13 14:39:55.279: ccsip_spi_process_event:Could not find ccb for response *Mar 13 14:39:58.551: Received: SIP/2.0 415 Unsupported Media Type Via: SIP/2.0/UDP 10.0.0.19:5060;x-route-tag="tgrp:Iskra" From: <sip:38513391900@10.0.0.19>;tag=2D49486C-1855 To: <sip:38514001148@89.18.32.19>;tag=ag2mFXH0NpctD Call-ID: 253738A2-F04211DC-81719B71-D67790FB@10.0.0.19 CSeq: 101 INVITE Timestamp: 1205419179 0.003927 User-Agent: FreeSWITCH-mod_sofia/1.0.pre3-7870 Accept: application/sdp Accept-Encoding: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO Supported: 100rel, precondition, timer Content-Length: 0 *Mar 13 14:39:58.555: HandleUdpSocketReads :Msg enqueued for SPI with IPaddr: 89.18.32.19:5080 *Mar 13 14:39:58.555: ccsip_spi_process_event:Could not find ccb for response *Mar 13 14:40:03.651: Received: SIP/2.0 415 Unsupported Media Type Via: SIP/2.0/UDP 10.0.0.19:5060;x-route-tag="tgrp:Iskra" From: <sip:38513391900@10.0.0.19>;tag=2D49486C-1855 To: <sip:38514001148@89.18.32.19>;tag=ag2mFXH0NpctD Call-ID: 253738A2-F04211DC-81719B71-D67790FB@10.0.0.19 CSeq: 101 INVITE Timestamp: 1205419179 0.003927 User-Agent: FreeSWITCH-mod_sofia/1.0.pre3-7870 Accept: application/sdp Accept-Encoding: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO Supported: 100rel, precondition, timer Content-Length: 0 *Mar 13 14:40:03.651: HandleUdpSocketReads :Msg enqueued for SPI with IPaddr: 89.18.32.19:5080 *Mar 13 14:40:03.655: ccsip_spi_process_event:Could not find ccb for response *Mar 13 14:40:06.663: Received: SIP/2.0 415 Unsupported Media Type Via: SIP/2.0/UDP 10.0.0.19:5060;x-route-tag="tgrp:Iskra" From: <sip:38513391900@10.0.0.19>;tag=2D49486C-1855 To: <sip:38514001148@89.18.32.19>;tag=ag2mFXH0NpctD Call-ID: 253738A2-F04211DC-81719B71-D67790FB@10.0.0.19 CSeq: 101 INVITE Timestamp: 1205419179 0.003927 User-Agent: FreeSWITCH-mod_sofia/1.0.pre3-7870 Accept: application/sdp Accept-Encoding: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO Supported: 100rel, precondition, timer Content-Length: 0 *Mar 13 14:40:06.663: HandleUdpSocketReads :Msg enqueued for SPI with IPaddr: 89.18.32.19:5080 *Mar 13 14:40:06.667: ccsip_spi_process_event:Could not find ccb for response *Mar 13 14:40:10.431: Received: SIP/2.0 415 Unsupported Media Type Via: SIP/2.0/UDP 10.0.0.19:5060;x-route-tag="tgrp:Iskra" From: <sip:38513391900@10.0.0.19>;tag=2D49486C-1855 To: <sip:38514001148@89.18.32.19>;tag=ag2mFXH0NpctD Call-ID: 253738A2-F04211DC-81719B71-D67790FB@10.0.0.19 CSeq: 101 INVITE Timestamp: 1205419179 0.003927 User-Agent: FreeSWITCH-mod_sofia/1.0.pre3-7870 Accept: application/sdp Accept-Encoding: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO Supported: 100rel, precondition, timer Content-Length: 0 *Mar 13 14:40:10.431: HandleUdpSocketReads :Msg enqueued for SPI with IPaddr: 89.18.32.19:5080 *Mar 13 14:40:10.431: ccsip_spi_process_event:Could not find ccb for response
Oh this bug has already been sent to MikeJ and Anthm. This is one strange invite. anyway its being worked on. You might want to see what in your config causes the addition of the application/gtd type. I suspect 'supplementary-service pass-through' /b INVITE sip:38514001148@89.18.32.19:5080 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.19:5060;x-route-tag="tgrp:Iskra" From: <sip:38513391900@10.0.0.19>;tag=2D49486C-1855 To: <sip:38514001148@89.18.32.19> Date: Thu, 13 Mar 2008 14:39:39 GMT Call-ID: 253738A2-F04211DC-81719B71-D67790FB@10.0.0.19 Supported: timer,100rel Min-SE: 1800 Cisco-Guid: 624255889-4030861788-2171509617-3598160123 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO CSeq: 101 INVITE Max-Forwards: 6 Remote-Party-ID: <sip: 38513391900@10.0.0.19>;party=calling;screen=yes;privacy=off Timestamp: 1205419179 Contact: <sip:38513391900@10.0.0.19:5060> Expires: 180 Allow-Events: telephone-event MIME-Version: 1.0 Content-Type: multipart/mixed;boundary=uniqueBoundary Content-Length: 523 --uniqueBoundary Content-Type: application/sdp v=0 o=CiscoSystemsSIP-GW-UserAgent 5947 1401 IN IP4 10.0.0.19 s=SIP Call c=IN IP4 10.0.0.19 t=0 0 m=audio 16850 RTP/AVP 8 c=IN IP4 10.0.0.19 a=rtpmap:8 PCMA/8000 a=ptime:20 --uniqueBoundary Content-Type: application/gtd Content-Disposition: signal;handling=optional IAM, PRN,isdn*,,NET5*, USI,rate,c,3,c,1 USI,lay1,alaw TMR,02 CPN,04,,1,14001148 CGN,04,,1,y,4,13391900 CPC,09 FCI,,,,,,,y, GCI,25356391f04211dc816e9b71d67790fb --uniqueBoundary-- On Mar 13, 2008, at 9:51 AM, Josip Djuricic wrote:
supplementary-service pass-through
I can disable GDP but still no audio, I'm checkin the traces now Brian West wrote:
Oh this bug has already been sent to MikeJ and Anthm. This is one strange invite. anyway its being worked on. You might want to see what in your config causes the addition of the application/gtd type. I suspect 'supplementary-service pass-through'
/b
INVITE sip:38514001148@89.18.32.19 <mailto:38514001148@89.18.32.19>:5080 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.19:5060;x-route-tag="tgrp:Iskra" From: <sip:38513391900@10.0.0.19 <mailto:38513391900@10.0.0.19>>;tag=2D49486C-1855 To: <sip:38514001148@89.18.32.19 <mailto:38514001148@89.18.32.19>> Date: Thu, 13 Mar 2008 14:39:39 GMT Call-ID: 253738A2-F04211DC-81719B71-D67790FB@10.0.0.19 <mailto:253738A2-F04211DC-81719B71-D67790FB@10.0.0.19> Supported: timer,100rel Min-SE: 1800 Cisco-Guid: 624255889-4030861788-2171509617-3598160123 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO CSeq: 101 INVITE Max-Forwards: 6 Remote-Party-ID: <sip:38513391900@10.0.0.19 <mailto:38513391900@10.0.0.19>>;party=calling;screen=yes;privacy=off Timestamp: 1205419179 Contact: <sip:38513391900@10.0.0.19 <mailto:38513391900@10.0.0.19>:5060> Expires: 180 Allow-Events: telephone-event MIME-Version: 1.0 Content-Type: multipart/mixed;boundary=uniqueBoundary Content-Length: 523 --uniqueBoundary Content-Type: application/sdp v=0 o=CiscoSystemsSIP-GW-UserAgent 5947 1401 IN IP4 10.0.0.19 s=SIP Call c=IN IP4 10.0.0.19 t=0 0 m=audio 16850 RTP/AVP 8 c=IN IP4 10.0.0.19 a=rtpmap:8 PCMA/8000 a=ptime:20 --uniqueBoundary Content-Type: application/gtd Content-Disposition: signal;handling=optional IAM, PRN,isdn*,,NET5*, USI,rate,c,3,c,1 USI,lay1,alaw TMR,02 CPN,04,,1,14001148 CGN,04,,1,y,4,13391900 CPC,09 FCI,,,,,,,y, GCI,25356391f04211dc816e9b71d67790fb --uniqueBoundary-- On Mar 13, 2008, at 9:51 AM, Josip Djuricic wrote:
supplementary-service pass-through
------------------------------------------------------------------------
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I meant GTD ;) Anyway, the traces now looks alot better, no SIP 415 anymore, but there is still no voice, and rtp seems to be going only from the phone to the FS Josip Djuricic wrote:
I can disable GDP but still no audio, I'm checkin the traces now
Brian West wrote:
Oh this bug has already been sent to MikeJ and Anthm. This is one strange invite. anyway its being worked on. You might want to see what in your config causes the addition of the application/gtd type. I suspect 'supplementary-service pass-through'
/b
INVITE sip:38514001148@89.18.32.19 <mailto:38514001148@89.18.32.19>:5080 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.19:5060;x-route-tag="tgrp:Iskra" From: <sip:38513391900@10.0.0.19 <mailto:38513391900@10.0.0.19>>;tag=2D49486C-1855 To: <sip:38514001148@89.18.32.19 <mailto:38514001148@89.18.32.19>> Date: Thu, 13 Mar 2008 14:39:39 GMT Call-ID: 253738A2-F04211DC-81719B71-D67790FB@10.0.0.19 <mailto:253738A2-F04211DC-81719B71-D67790FB@10.0.0.19> Supported: timer,100rel Min-SE: 1800 Cisco-Guid: 624255889-4030861788-2171509617-3598160123 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO CSeq: 101 INVITE Max-Forwards: 6 Remote-Party-ID: <sip:38513391900@10.0.0.19 <mailto:38513391900@10.0.0.19>>;party=calling;screen=yes;privacy=off Timestamp: 1205419179 Contact: <sip:38513391900@10.0.0.19 <mailto:38513391900@10.0.0.19>:5060> Expires: 180 Allow-Events: telephone-event MIME-Version: 1.0 Content-Type: multipart/mixed;boundary=uniqueBoundary Content-Length: 523 --uniqueBoundary Content-Type: application/sdp v=0 o=CiscoSystemsSIP-GW-UserAgent 5947 1401 IN IP4 10.0.0.19 s=SIP Call c=IN IP4 10.0.0.19 t=0 0 m=audio 16850 RTP/AVP 8 c=IN IP4 10.0.0.19 a=rtpmap:8 PCMA/8000 a=ptime:20 --uniqueBoundary Content-Type: application/gtd Content-Disposition: signal;handling=optional IAM, PRN,isdn*,,NET5*, USI,rate,c,3,c,1 USI,lay1,alaw TMR,02 CPN,04,,1,14001148 CGN,04,,1,y,4,13391900 CPC,09 FCI,,,,,,,y, GCI,25356391f04211dc816e9b71d67790fb --uniqueBoundary-- On Mar 13, 2008, at 9:51 AM, Josip Djuricic wrote:
supplementary-service pass-through
------------------------------------------------------------------------
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I dont know how relevant this is but from my personal experience with Asterisk and Cisco When I use insecure=invite or very i get issues but if i use insecure=yes it works fine On Thu, Mar 13, 2008 at 10:12 AM, Josip Djuricic <josip.djuricic@primatel.hr> wrote:
Hi there,
I was already writing about this problem, and I still didn't managed to fix it. I thought it was a NAT problem, but now without NAT the problem still remains.
The error is like this, when I call from pstn, the call goes trough Cisco AS5300, then goes to FS, then to it's final destination - Ip Phone.
Now you can answer the phone, but you can't hear anything on any side. And FS returns lots of SIP 415 (unsupported media type) to Cisco AS5300.
When you call from Voip phone to the same pstn number it all works well, so I'm starting to be confused more and more. (and in this case there is no 415 messages).
I believe cisco is sending something that FS can't understand, in the INVITE's but that's just a wild guess cause I ain't sure what can FS exactly understand very well.
Whoever want's I can send him a tcpdump, it's about 7mb tarred, so I can't send it to the list.
Thank you all,
Josip
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participants (5)
-
Brian West
-
David Knell
-
Josip Djuricic
-
Tamas
-
Vijay Ramnarine