Dear Nipun and Narinder,
Sorry that I did not answer before. We had an external guy here yesterday for another topic and we used the time and asked him regarding this issue.
Before this, we checked if we had any slips. And we really had some Slip secs. So we played a bit with the clock configuration until this was stable.
Then we tested:
dial-peer voice 500 voip fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711ulaw fax-relay ecm disable
fax nsf 000000 fax-relay sg3-to-g3
This did also not work.
Then the external had the idea to change the fax protocol with version 3.
So now the dialpeer looks like this:
dial-peer voice 500 voip description SIP to CUCM preference 1 destination-pattern +350200025.. session protocol sipv2 session transport tcp session server-group 100 voice-class codec 10 voice-class sip options-keepalive profile 100 voice-class sip bind control source-interface Loopback0 voice-class sip bind media source-interface Loopback0 dtmf-relay rtp-nte sip-notify fax-relay ecm disable fax-relay sg3-to-g3 fax rate 9600 fax nsf 000000 fax protocol t38 version 3 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711ulaw no vad
And this FINALLY WORKS! Incredible how hard this was!
Thanks to both of you for all the support! :)
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Thanks for your replies!
Okay, as you say there is really something strange since the gateway is not replying to the first connection request.
Narinder, I've added the command voice iec syslog to the global config.
Attached are the Sh Run and sh version. And also a call trace to the fax with the following debug commands:
- debug ccsip message - debug ccsip error - debug isdn q931 - debug voip vtsp all - debug voice ccapi inout
Thanks for helping!
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Dear Nipun and Narinder,
First, thank you for your responses and your time.
Nipun, to your questions:
- Good point! I've done a test last friday to a normal phone and we get the same q931 error! But it works anyway. So we can heanr and the other side hears us also. So the issue is happening not only to rightfax.
- The RightFax is connected to the Call Manager directly with a Sip Trunk. We have many many faxes working over this without issues. Only Gibraltar has the problem.
- Correct, the destination of the CUCM->Gateway is to the Loopback address.
Narinder, to your questions:
- Correct, the 503 Response is received by the Gateway from CUCM.
- I've uploaded a complete trace from the Gateway calling a Phone (2523)and calling a Fax (2501).
- Correct. That is the route the Call does - PSTN->gateway->sip cucm-> cucm sip to fax server
- Yes I can see traffic outgoing and incoming from the CUCM to the RightFax. As I said before it's working for many other sites.
- "Options Ping" is not enabled.
The Traces are attached. Hopefully you can see something.
The logs are from external mobile phone to the Fax and to a Phone in Gib:
- debug isdn error
- debug isdn q931
- sh cssip messages all
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If I only do a "debug isdn q931" I get this message:
Apr 6 09:41:53.717: ISDN Se0/1/0:15 Q931: RX <- SETUP pd = 8 callref = 0x0001 Sending Complete Bearer Capability i = 0x8090A3 Standard = CCITT Transfer Capability = Speech Transfer Mode = Circuit Transfer Rate = 64 kbit/s Channel ID i = 0xA98381 Exclusive, Channel 1 Progress Ind i = 0x8381 - Call not end-to-end ISDN, may have in-band info Progress Ind i = 0x8483 - Origination address is non-ISDN Calling Party Number i = 0x1183, '41797329888' Plan:ISDN, Type:International Called Party Number i = 0xC1, '02501' Plan:ISDN, Type:Subscriber(local) Apr 6 09:41:53.717: ISDN Se0/1/0:15 Q931: Received SETUP callref = 0x8001 callID = 0x01E7 switch = primary-net5 interface = User Apr 6 09:41:53.724: ISDN Se0/1/0:15 Q931: TX -> CALL_PROC pd = 8 callref = 0x8001 Channel ID i = 0xA98381 Exclusive, Channel 1 Apr 6 09:41:53.806: ISDN Se0/1/0:15 Q931: TX -> ALERTING pd = 8 callref = 0x8001 Apr 6 09:41:53.854: ISDN **ERROR**: validate_connected_number: Invalid connected_number Apr 6 09:41:53.854: ISDN Se0/1/0:15 Q931: TX -> CONNECT pd = 8 callref = 0x8001
I've checked the translations and even the traffic. It's a RightFax. I can see traffic from the Gateway to the Rightfax server. All looks really good. Just the problem that the fax is muted and cannot receive faxes like this.
Thanks for any help.
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We have replaced an old voice gateway Cisco 2811 with a new Cisco 4331 and "ported" more or less all relevant configuration from the old to the new.
All Calls incoming and outgoing are working fine to our phones. But the incoming Fax from external are not working.
It rings 1 time (arrives at our gateway) and then we hear nothing. The call stays connected but there is no fax sound as we normally expect.
If I call the fax internally (not going over Gateway), the fax does its well known sound and works.
And with the old gateway it also works.
Fax Number: +35020002501
I've done many tests and configuration changes to test but had no luck till now. Below you can see all my debugs:
sh voice call status CallID CID ccVdb Port Slot/Bay/DSP:Ch Called # Codec MLPP Dial-peers 0x6E0EC 1388 0x7FFDA96E1740 0/1/0:15.1 1/1:1 5020002501 g711ulaw 10/500 1 active call found
Extract of the configuration:
voice service voip ip address trusted list ipv4 Server1 ipv4 Server2 ipv4 Server3 ipv4 Server4 allow-connections sip to sip no supplementary-service sip handle-replaces fax protocol t38 version 0 ls-redundancy 2 hs-redundancy 0 fallback pass-through g711ulaw modem passthrough nse codec g711ulaw sip bind control source-interface Loopback0 bind media source-interface Loopback0 registrar server no update-callerid ! voice class codec 10 codec preference 1 g711ulaw codec preference 2 g711alaw codec preference 3 g729r8 ! ! voice class sip-profiles 1 request INVITE sip-header From modify "<sip:anonymous@" "<sip:" request INVITE sip-header From modify "anonymous" "" ! ! voice class server-group 100 ipv4 Server1 preference 1 ipv4 Server2 preference 2 ipv4 Server3 preference 3 ipv4 Server4 preference 4 description UCM Server Group !
! voice translation-rule 20 rule 1 /^2000\(....\)$/ /+3502000\1/ rule 2 /^025\(..\)$/ /+350200025\1/ ! voice translation-rule 21 rule 1 /^/ /+35020/ type subscriber subscriber rule 2 /^/ /+350/ type national national rule 3 /^/ /+/ type international international !
! voice translation-profile PSTN2ITN translate calling 21 translate called 20 !
! dial-peer voice 10 pots description PSTN Incoming Dialpeer translation-profile incoming PSTN2ITN call-block translation-profile incoming blacklisted-calls call-block disconnect-cause incoming call-reject preference 1 incoming called-number .T direct-inward-dial port 0/1/0:15 !
! dial-peer voice 500 voip description SIP to CUCM preference 1 destination-pattern +350200025.. session protocol sipv2 session transport tcp session server-group 100 voice-class codec 10 voice-class sip options-keepalive profile 100 voice-class sip bind control source-interface Loopback0 voice-class sip bind media source-interface Loopback0 dtmf-relay rtp-nte sip-notify fax-relay sg3-to-g3 no vad !
Extract of the relevant logs I've found.
Right before any Logs with the Calling and Called numbers show up I get this message:
Received: SIP/2.0 503 Service Unavailable Via: SIP/2.0/TCP ***GATEWAY LOOPBACK***:5060;branch=z9hG4bK44BE5D50 From: <sip:***GATEWAY LOOPBACK***>;tag=6111E241-D7C To: <sip:***PUBLISHER IPADDRESS***>;tag=1149161109 Date: Wed, 04 Apr 2018 09:02:59 GMT Call-ID: CE5B75F8-371D11E8-A979AD25-DD7BDD2E@10.50.66.1 CSeq: 101 OPTIONS Warning: 399 ***PUBLISHER HOSTNAME*** "Unable to find a device handler for the request received on port 51036 from ***GATEWAY LOOPBACK***" Content-Length: 0
This are the next logs:
Apr 4 11:03:04.343: ISDN Se0/1/0:15 Q931: RX <- SETUP pd = 8 callref = 0x0001 Sending Complete Bearer Capability i = 0x9090A3 Standard = CCITT Transfer Capability = 3.1kHz Audio Transfer Mode = Circuit Transfer Rate = 64 kbit/s Channel ID i = 0xA98381 Exclusive, Channel 1 Progress Ind i = 0x8381 - Call not end-to-end ISDN, may have in-band info Progress Ind i = 0x8483 - Origination address is non-ISDN Calling Party Number i = 0x1183, '41797329xxx' *** My mobile number Plan:ISDN, Type:International Called Party Number i = 0xC1, '02501' Plan:ISDN, Type:Subscriber(local) Apr 4 11:03:04.343: ISDN Se0/1/0:15 Q931: Received SETUP callref = 0x8001 callID = 0x01C0 switch = primary-net5 interface = User
Dial Peer matching....
Apr 4 11:03:04.353: //447699/D183693D818E/SIP/Info/critical/32768/ccsip_ipip_media_forking_update_preferred_codec: MF: Not a Forked SIP leg.. Apr 4 11:03:04.353: //447699/D183693D818E/SIP/Info/info/8704/sipSPIGetCallConfig: Incoming: No defer BYE for last call stats Apr 4 11:03:04.353: //447699/D183693D818E/SIP/Info/verbose/1/ccsip_set_srtp_config: No Srtp configure for this leg. Apr 4 11:03:04.353: //447699/D183693D818E/SIP/Info/info/8192/sipSPIGetCallConfig: Media forking disabled Apr 4 11:03:04.353: //447699/D183693D818E/SIP/Info/info/34816/ccsip_ipip_media_forking_anchor_leg_config: MF: en_p->encap_s.voIP.voipPeerCfgMediaClass = 0 Apr 4 11:03:04.353: //447699/D183693D818E/SIP/Info/info/34816/ccsip_ipip_media_forking_anchor_leg_config: MF: Dial-peer has no media class recorder. Apr 4 11:03:04.353: //447699/D183693D818E/SIP/Info/info/36864/sipSPIMFChangeState: MF: Prev state = 0 & New state = -1 Apr 4 11:03:04.353: //447699/D183693D818E/SIP/Info/info/32768/ccsip_ipip_media_forking_anchor_leg_reset: MF: Anchor leg config reset done... Apr 4 11:03:04.353: //447699/D183693D818E/SIP/Info/info/32768/ccsip_ipip_media_forking_intra_frame_request_config: MF: FIR en_p->encap_s.voIP.voipPeerCfgMediaClass = 0 Apr 4 11:03:04.353: //447699/D183693D818E/SIP/Info/info/32768/ccsip_ipip_media_forking_get_forked_leg_config: MF: This leg is not forked call leg. Apr 4 11:03:04.353: //447699/D183693D818E/SIP/Info/critical/11264/ccsipInitDSCPPolicyInfo: No DSCP Profile configured, No RPH 2 DSCP Mapping and DSCP policing Apr 4 11:03:04.353: //447699/D183693D818E/SIP/Info/verbose/8192/sipSPIGetCallConfig: Initilise the DSCP policy Apr 4 11:03:04.353: //447699/D183693D818E/SIP/Info/verbose/8192/sipSPICheckFAAnatAssymetricOrDO2EO: Not a SIP-SIP call or not in FA mode Apr 4 11:03:04.353: //447699/D183693D818E/SIP/Info/notify/2049/populate_vcc_data: Using Voice Class Codec, tag = 10 and offer-all is = FALSE Apr 4 11:03:04.353: //447699/D183693D818E/SIP/Info/notify/8192/sipSPISetOverlapConfiguration: Overlap signaling: FALSE: Endpt: SIP Trunk Apr 4 11:03:04.353: //447699/D183693D818E/SIP/Info/verbose/10240/sipSPI_ipip_GetHdrPassthruCfg: Hdr passthrough config:1 tag:0 Apr 4 11:03:04.353: //447699/D183693D818E/SIP/Info/verbose/2048/sipSPI_ipip_GetCopyListCfg: Copy-list config:2 tag:0 Apr 4 11:03:04.353: //447699/D183693D818E/SIP/Info/info/2048/sipSPIGetExtensionCfg: SIP extension config:1, check sys cfg:1 Apr 4 11:03:04.353: //447699/D183693D818E/SIP/Info/notify/10240/sipSPI_ipip_build_consolidated_header_list: Both passthru and copylist are disabled Apr 4 11:03:04.353: //447699/D183693D818E/SIP/Info/info/1/preprocessSetup: This is a not a SIGO Call -, could be DM call Apr 4 11:03:04.354: //447699/D183693D818E/SIP/State/ccsip_cnfsm_debugs: IWF:cur_container:sip_iwf_default_early_dialog_container, cur_state:S_SIP_IWF_SDP_IDLE, event:E_SIP_IWF_EV_INIT_CALL_SETUP Apr 4 11:03:04.354: //447699/D183693D818E/SIP/State/ccsip_cnfsm_debugs: IWF:new_container:sip_iwf_main_container Apr 4 11:03:04.354: //447699/D183693D818E/SIP/Info/verbose/4096/ccsip_iwf_process_event: IWF - cnfsm ret 2 Apr 4 11:03:04.354: //447699/D183693D818E/SIP/Info/notify/4096/preprocessSetup: SIP-TDM or TCL/VXML app case Apr 4 11:03:04.354: //447699/D183693D818E/SIP/Info/notify/6/sipSPIValidateStreamAddrType: stream:1, Mode : 1 Apr 4 11:03:04.354: //447699/D183693D818E/SIP/Info/verbose/513/resolve_media_ip_address_to_bind: peer_tag=500 Apr 4 11:03:04.354: //-1/xxxxxxxxxxxx/SIP/Info/info/8192/resolve_media_ip_address_to_bind: VRF id = 0 Apr 4 11:03:04.354: //-1/xxxxxxxxxxxx/SIP/Info/info/8192/ccsip_get_ifaddress: ip_address IPv4 ***GATEWAY LOOPBACK*** for SIP Apr 4 11:03:04.354: //-1/xxxxxxxxxxxx/SIP/Info/info/8192/resolve_ip_address_to_bind: ip_get_ifaddress IPv4 ***GATEWAY LOOPBACK*** for SIP Apr 4 11:03:04.354: //447699/D183693D818E/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = ***GATEWAY LOOPBACK*** Apr 4 11:03:04.354: //447699/D183693D818E/SIP/Info/critical/1/sipSPIOutgoingCallSDP: Failure in creating outbound streams SIP: (447699) Group (a= group line) attribute, level 65535 instance 1 not found. Apr 4 11:03:04.354: //447699/D183693D818E/SIP/Info/info/8192/resolve_sig_ip_address_to_bind: VRF id = 0 Apr 4 11:03:04.354: //-1/xxxxxxxxxxxx/SIP/Info/info/8192/ccsip_get_ifaddress: ip_address IPv4 ***GATEWAY LOOPBACK*** for SIP Apr 4 11:03:04.354: //-1/xxxxxxxxxxxx/SIP/Info/info/8192/resolve_ip_address_to_bind: ip_get_ifaddress IPv4 ***GATEWAY LOOPBACK*** for SIP Apr 4 11:03:04.354: //447699/D183693D818E/SIP/Info/info/8192/resolve_sig_ip_address_to_bind: signaling bind address : ***GATEWAY LOOPBACK*** Apr 4 11:03:04.354: //-1/xxxxxxxxxxxx/SIP/Info/info/8192/resolve_sig_ip_address_to_bind: return addr ***GATEWAY LOOPBACK*** Apr 4 11:03:04.354: //-1/xxxxxxxxxxxx/SIP/Media/sipSPIReserveRtpPort: reserved port 19828 for stream 1 Apr 4 11:03:04.354: //447699/D183693D818E/SIP/Info/info/1/sipSPIDoBearerCapToCodecMapping: Bearer capability to Codec Mapping: DISABLED
Apr 4 11:03:04.354: //447699/D183693D818E/SIP/Media/sipSPIAddSDPMediaPayload: Preferred method of dtmf relay is: 6, with payload: 101 Apr 4 11:03:04.354: //447699/D183693D818E/SIP/Info/info/131074/sipSPIBwCacCalcMaxAudioBandwidth: calculating max bw from preffered codecs (local offer) SIP: (447699) Group (a= group line) attribute, level 65535 instance 1 not found. Apr 4 11:03:04.354: //447699/D183693D818E/SIP/Info/info/131074/sipSPIBwCacCalcMaxAudioBandwidth: max bw (excluding pak overhead) from preffered codecs: codec g711ulaw bw 64000 index 0 Apr 4 11:03:04.354: //447699/D183693D818E/SIP/Info/critical/2/sipSPIBwCacCalcMaxAudioBandwidth: audio caps channel idx not found !!!! Apr 4 11:03:04.354: //-1/xxxxxxxxxxxx/SIP/Info/notify/1/convert_codec_bytes_to_ptime: Values :Codec: g711ulaw codecbytes :160, ptime: 20 Apr 4 11:03:04.442: //-1/xxxxxxxxxxxx/SIP/Info/critical/1024/httpish_msg_process_network_msg: HEADER LINE READ FAILURE DUE TO RS->EOF Apr 4 11:03:04.442: //-1/xxxxxxxxxxxx/SIP/Info/critical/1024/ccsip_process_network_message: process_network_msg: not complete Apr 4 11:03:04.442: //-1/xxxxxxxxxxxx/SIP/Info/info/1024/httpish_msg_free: Freed msg=0x7FFDB30983B0 Apr 4 11:03:04.442: //-1/xxxxxxxxxxxx/SIP/Info/critical/4096/sip_tcp_newmsg_to_spi: process_network_msg: not complete Apr 4 11:03:04.442: //-1/xxxxxxxxxxxx/SIP/Info/info/1024/httpish_msg_free: Freed msg=0x7FFDB3098A28 Apr 4 11:03:04.484: //-1/xxxxxxxxxxxx/SIP/Transport/sip_find_connid_by_fd: Map fd 7 to index 65
I tried to search this error messages but had no luck.
Do you know anything I could try to solve this issue? Or do you need more debugs?
Thanks all for any help!
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