Just to re-cap (it's been a few days!) i'm using Cisco 7960's and have tried setting both phones to different codecs (tried default g729a, g711alaw, and g711ulaw). Also, the other observations that have been made:
- Problem is one-way. One side hears me clearly ; I don't hear the other side clearly at all (5% audible only).
- Calls to MSN are fine (two way conversation is crystal clear)
- Calls to a Zultys Zip2 SIP phone is also perfectly clear.
- All these three tested over the same network and same VPN (call between Hong Kong and USA).
- Cisco to Cisco calls worked fine with Vocal.
If Cisco is able to talk fine with other devices, there should not be a problem with bandwidth or my network. However, I am finding it quite bizzarre that Cisco is unable to talk to itself. The problem shouldn't be VAD or the like - even if I talk non-stop, or the other guy does, I get the same problem.
I attach a copy of my Cisco phone configuration for reference. I have even recently upgraded my phone firmware - but no luck.
Platform : Cisco IP Phone 7960 Elasped Time: 08:11:26
dhcp_server : 192.168.8.254 my_ip_addr : 192.168.8.83 subnet_mask : 255.255.255.0 defaultgw : 192.168.8.254 dyn_dns_addr_1 : 0.0.0.0 dyn_dns_addr_2 : 0.0.0.0 dns_addr : 205.252.144.228 dns_backup_1: 202.14.67.4 tftp_addr : 192.168.0.252 dyn_tftp_addr : 0.0.0.0 my_mac_addr : 0007:50ac:6932 domain_name : deltapath.com my_name : SIP000750AC6932 Status Flags : 12300000
image_version : "P0S3-05-3-00" FirmLoadID : "PC03A300" network_media_type : Auto network_port2_type : Hub/Switch tos_media : 5 phone_label : "DELTAPATH" tftp_cfg_dir : "./sip_phone/" phone_password : ********** phone_prompt : "SIP Phone" language : english sntp_mode : DirectedBroadcast sntp_server : stdtime.gov.hk time_zone : HST dst_offset : 0 dst_start_month : April dst_start_day : 0 dst_start_day_of_week : Sun dst_start_week_of_month : 1 dst_start_time : 02 dst_stop_month : Oct dst_stop_day : 0 dst_stop_day_of_week : Sunday dst_stop_week_of_month : 8 dst_stop_time : 2 dst_auto_adjust : 0 time_format_24hr : 1 date_format : M/D/Y nat_enable : 0 nat_address : voip_control_port : 5060 start_media_port : 16384 end_media_port : 32766 sync : "1" xml_card_dir : "" xml_card_file : "CARD.XML" telnet_level : 2 services_url : "" directory_url : "" logo_url : "http://deltapath.com/logo.bmp"; http_proxy_addr : http_proxy_port : 80 enable_vad : 0 dial_template : "dialplan" callerid_blocking : 0 anonymous_call_block : 0 autocomplete : 1 messages_uri : "86" dnd_control : 0 preferred_codec : g729a dtmf_outofband : avt dtmf_avt_payload : 101 dtmf_db_level : 3 dtmf_inband : 1 line1_name : "TerenceParker" line2_name : "74xxx" line3_name : "74xxx" line4_name : "" line5_name : "" line6_name : "" line1_authname : "TerenceParker" line2_authname : "74xxx" line3_authname : "74xxx" line4_authname : "UNPROVISIONED" line5_authname : "UNPROVISIONED" line6_authname : "UNPROVISIONED" line1_shortname : "Asterisk" line2_shortname : "FWD-74xxx" line3_shortname : "FWD-74xxx" line4_shortname : "UNPROVISIONED" line5_shortname : "UNPROVISIONED" line6_shortname : "UNPROVISIONED" line1_displayname : "TerenceParker" line2_displayname : "74xxx" line3_displayname : "Terence Parker" line4_displayname : "" line5_displayname : "" line6_displayname : "" proxy1_address : "192.168.0.254" proxy2_address : "fwd.pulver.com" proxy3_address : "fwd.pulver.com" proxy4_address : "" proxy5_address : "" proxy6_address : "" proxy1_port : 5060 proxy2_port : 5060 ........ sip_retx : 10 sip_invite_retx : 6 timer_t1 : 500 timer_t2 : 4000 timer_invite_expires : 180 timer_register_expires : 3600 proxy_register : 1 proxy_backup : "UNPROVISIONED" proxy_emergency : "UNPROVISIONED" proxy_backup_port : 0 proxy_emergency_port : 0 outbound_proxy : outbound_proxy_port : 5082 nat_received_processing : 0 mwi_status : 0 call_waiting : 1 user_info : none cnf_join_enable : 1 remote_party_id : 0 semi_attended_transfer : 1 call_hold_ringback : 0
Thanks for any help!
Terence
I have never used Cisco phones, but I have had problems in the past relating to * RTP talking to a widget with VAD turned on. * RTP stack can not run on its own. It relies on receiving RTP packets for doing its timing.
A simple test is to sniff the line to make sure the phones always send packets.
If you see pauses, you may need to disable some type of VAD setting on the phone.
Or just never quit talking when using the Cisco phone.
Terence Parker wrote:
I have set canreinvite=no in the sip.conf for each user (well, there are
only two) using a cisco phone. What does this imply?
As for whether the problem is due to the phones or asterisk however, indications would suggest both, because:
- Voicemail works fine (and is clear)
- I can initiate a call between MSN and Cisco, and that would sound fine.
This might suggest a problem with my phones. However :
- When using Vocal previously, Cisco to Cisco conversation was fine.
This has led me to be completely stumped! I notice some mention elsewhere
about asterisk lacking certain codecs because of license restrictions? Is
this anything to do with me? Or should the phones still - in theory - be
able to talk to each other without any problems? I have tried the cisco
phone on both g729a and g711ulaw.
I'm currently *trying* to get ahold of an updated firmware for my phone. I
will see if this fixes the problems.
Thanks again,
Terence
_______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
![http://deltapath.com/logo.bmp"](http://deltapath.com/logo.bmp"){kind=link}